---
[2011-11-03 06:46:01]   == Extension Changed 4113[default-local] new state Unavailable for Notify User 4124
[2011-11-03 06:46:01] set_destination: Parsing <sip:4149@172.30.0.188:5060> for address/port to send to
[2011-11-03 06:46:01] set_destination: set destination to 172.30.0.188:5060
[2011-11-03 06:46:01] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]   == Extension Changed 4113[default-local] new state Unavailable for Notify User 4149
[2011-11-03 06:46:01]   == Extension Changed 4113[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:01] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:46:01] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:46:01] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3472001b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as2fb231f0
To: "4148" <sip:4148@customer.com>;tag=EE625669-435859C8
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: af04dc4c-fc6ea483-38848532@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]   == Extension Changed 4113[default-local] new state Unavailable for Notify User 4148
[2011-11-03 06:46:01] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:46:01] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:46:01] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a8e32c8
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7ac82345
To: "Office 106" <sip:4120@customer.com>;tag=3F364E51-5B222F0
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 9f66077-d16d29c6-d057424d@172.30.0.180
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]   == Extension Changed 4113[default-local] new state Unavailable for Notify User 4120
[2011-11-03 06:46:01] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:01] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:01] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54c8630b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as4f80e97d
To: "4136" <sip:4136@customer.com>;tag=EBBF8BD9-54CCB672
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: bef4b4f0-29e154c9-65f17362@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]   == Extension Changed 4113[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:01] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:46:01] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:46:01] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2a10e6db
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as15700ff6
To: "Leon Zhong" <sip:4134@customer.com>;tag=32A0D36-967AF257
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 7cdaf61-6dca67e6-6d8a2347@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]   == Extension Changed 4113[default-local] new state Unavailable for Notify User 4134
[2011-11-03 06:46:01]
<--- SIP read from UDP:172.30.0.168:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5cf359f0
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
CSeq: 105 NOTIFY
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Content-Length: 0


<------------->
[2011-11-03 06:46:01] --- (10 headers 0 lines) ---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK378ea673
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6e02a2c0
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 039134d54c922e5279d167e65f157b75@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c360bcd
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as45b9714a
To: "Office 108" <sip:4128@customer.com>;tag=BFDCB33E-CD5B5BA7
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 1f0e8cd6-f0b74f7f-c83fed80@172.30.0.182
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK266230cf
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as7c6d37d6
To: "4147" <sip:4147@customer.com>;tag=4C1D584F-2F9D61D0
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 3af79169-77fdc85a-8cf41d63@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e45588d
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b008ef1
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as08b6f44c
To: "4124" <sip:4124@customer.com>;tag=DAA75691-6EF42394
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: b5bf6aea-f1839b95-b22ea038@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22f715bb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08ad6527
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2e051206521ace410e55bcfd14ccabc3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:01] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3472001b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as2fb231f0
To: "4148" <sip:4148@customer.com>;tag=EE625669-435859C8
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: af04dc4c-fc6ea483-38848532@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a8e32c8
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7ac82345
To: "Office 106" <sip:4120@customer.com>;tag=3F364E51-5B222F0
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 9f66077-d16d29c6-d057424d@172.30.0.180
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4115@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKeded657e853B8849
From: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
To: <sip:4115@customer.com>;tag=as4b34e14d
CSeq: 3 SUBSCRIBE
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4140", realm="asterisk", nonce="010e5ea6", uri="sip:4115@customer.com:5060", response="39a89ae57033ce63cb910c4b21a61a5e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:01] --- (16 headers 0 lines) ---
[2011-11-03 06:46:01] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:46:01] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4140" <sip:4140@customer.com>;tag=AA667848-38F99023'
[2011-11-03 06:46:01]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKeded657e853B8849;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
To: <sip:4115@customer.com>;tag=as4b34e14d
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="093cc780", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:01] Scheduling destruction of SIP dialog '9be0f5d8-a82d0833-1c4d0606@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:01] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54c8630b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as4f80e97d
To: "4136" <sip:4136@customer.com>;tag=EBBF8BD9-54CCB672
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: bef4b4f0-29e154c9-65f17362@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2a10e6db
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as15700ff6
To: "Leon Zhong" <sip:4134@customer.com>;tag=32A0D36-967AF257
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 7cdaf61-6dca67e6-6d8a2347@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5d84ee
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0f964613
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4bfef01f561aa11355bf50266675edc4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:01] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0975328c
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as728e16e0
To: "4136" <sip:4136@customer.com>;tag=19F02E68-92FA0841
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: cc56bb5f-1e5b7278-c0f81851@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5438a269
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as70865b2e
To: "Office 131" <sip:4113@customer.com>;tag=16D6ACA6-61364B5D
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: 2b297ea3-c6eac54a-1a74dcc1@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4135@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK1e8eb874F27FF219
From: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
To: <sip:4135@customer.com>;tag=as48c3b1a3
CSeq: 3 SUBSCRIBE
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="269fb622", uri="sip:4135@customer.com:5060", response="322e800edd37b088cf5352b23b8aebb9", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:01] --- (16 headers 0 lines) ---
[2011-11-03 06:46:01] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:46:01] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669'
[2011-11-03 06:46:01]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK1e8eb874F27FF219;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
To: <sip:4135@customer.com>;tag=as48c3b1a3
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1bb3528f", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:01] Scheduling destruction of SIP dialog '945e8061-662c26ca-2cbdbea7@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:01] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5c031c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5fb2bd9e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as79f82d48
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2f435bb9010529d15021f9d34b59a20e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:01]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6af06c6c
From: <sip:4124@customer.com>;tag=as0e065474
To: "4140" <sip:4140@customer.com>;tag=8DFECB39-66B5887C
CSeq: 104 NOTIFY
Call-ID: 14e5ed03-ab243756-85cfcde1@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:01] --- (10 headers 0 lines) ---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e45588d
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK31412891
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as37d895a1
To: "4148" <sip:4148@customer.com>;tag=C43C5186-E341D80D
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: ef0ee656-7679775d-5bbfcb5c@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b008ef1
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as08b6f44c
To: "4124" <sip:4124@customer.com>;tag=DAA75691-6EF42394
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: b5bf6aea-f1839b95-b22ea038@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3472001b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as2fb231f0
To: "4148" <sip:4148@customer.com>;tag=EE625669-435859C8
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: af04dc4c-fc6ea483-38848532@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a8e32c8
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7ac82345
To: "Office 106" <sip:4120@customer.com>;tag=3F364E51-5B222F0
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 9f66077-d16d29c6-d057424d@172.30.0.180
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54c8630b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as4f80e97d
To: "4136" <sip:4136@customer.com>;tag=EBBF8BD9-54CCB672
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: bef4b4f0-29e154c9-65f17362@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2a10e6db
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as15700ff6
To: "Leon Zhong" <sip:4134@customer.com>;tag=32A0D36-967AF257
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 7cdaf61-6dca67e6-6d8a2347@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Retransmitting #6 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7a7c8ee1
Max-Forwards: 70
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
Contact: <sip:4118@172.30.0.10:5060>
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4118">
<address uri="sip:4118@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK671c06bc
From: <sip:4145@customer.com>;tag=as1ab59962
To: "Office 107" <sip:4141@customer.com>;tag=DC2068B4-849F3759
CSeq: 103 NOTIFY
Call-ID: 178cff88-a06977bd-e4d9ec56@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:01] --- (10 headers 0 lines) ---
[2011-11-03 06:46:01] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:01]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.181:5060
[2011-11-03 06:46:01] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c360bcd
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as45b9714a
To: "Office 108" <sip:4128@customer.com>;tag=BFDCB33E-CD5B5BA7
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 1f0e8cd6-f0b74f7f-c83fed80@172.30.0.182
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:01] Really destroying SIP dialog '178cff88-a06977bd-e4d9ec56@172.30.0.181' Method: SUBSCRIBE
[2011-11-03 06:46:01]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6559f85e
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
CSeq: 104 NOTIFY
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:01] --- (10 headers 0 lines) ---
[2011-11-03 06:46:01] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:01]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.165:5060
[2011-11-03 06:46:01] Retransmitting #5 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04457a9c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5cff5298
To: "4168" <sip:4168@customer.com>;tag=530ADCB5-91794658
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 59688f55-c01b9f8-f830e953@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
[2011-11-03 06:46:01]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0eb61b61
From: <sip:4118@customer.com>;tag=as7cecdea0
To: "4168" <sip:4168@customer.com>;tag=44916A20-951EF5BB
CSeq: 103 NOTIFY
Call-ID: ae31c99c-fff7b197-b71cfe0a@172.30.0.187
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:01] --- (10 headers 0 lines) ---
[2011-11-03 06:46:01] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:01]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.187:5060
[2011-11-03 06:46:01] Retransmitting #6 (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK402808bc
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4b66acb0
To: "Office 105" <sip:4131@customer.com>;tag=4606BFFE-5F8266C9
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 3000b8a7-86241b9a-f1b5fbc5@172.30.0.170
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb9575a4448DC719B
From: "Office 131" <sip:4113@customer.com>;tag=BB7BAF6B-ECA88F92
To: <sip:4148@customer.com>;tag=as032151df
CSeq: 4 SUBSCRIBE
Call-ID: b9737532-4c0cc29-41dfcaf0@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="34aca8b9", uri="sip:4148@172.30.0.10:5060", response="22280b2fd7961f8c7049946a8ebd7701", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (16 headers 0 lines) ---
[2011-11-03 06:46:02] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:02]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb9575a4448DC719B;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=BB7BAF6B-ECA88F92
To: <sip:4148@customer.com>;tag=as032151df
Call-ID: b9737532-4c0cc29-41dfcaf0@172.30.0.195
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:02] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3000b8a7-86241b9a-f1b5fbc5@172.30.0.170 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6391ms with no response
[2011-11-03 06:46:02] Really destroying SIP dialog 'b9737532-4c0cc29-41dfcaf0@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4137@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKe0a5188aF0CCC825
From: "4115" <sip:4115@customer.com>;tag=F0659C17-802FEE32
To: <sip:4137@customer.com>;tag=as544277d6
CSeq: 4 SUBSCRIBE
Call-ID: 55f60a00-130f7c9b-436d40b6@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="2a566ac0", uri="sip:4137@172.30.0.10:5060", response="3ea8b42bde6aba215395333106eb5646", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (16 headers 0 lines) ---
[2011-11-03 06:46:02] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:02]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKe0a5188aF0CCC825;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=F0659C17-802FEE32
To: <sip:4137@customer.com>;tag=as544277d6
Call-ID: 55f60a00-130f7c9b-436d40b6@172.30.0.197
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e45588d
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b008ef1
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as08b6f44c
To: "4124" <sip:4124@customer.com>;tag=DAA75691-6EF42394
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: b5bf6aea-f1839b95-b22ea038@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #5 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK610ec4e0
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as3b261357
To: "Office 113" <sip:4171@customer.com>;tag=DE9E2A48-9DE3D0C3
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: e69ec24d-a1b6e62c-e18efe77@172.30.0.165
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3472001b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as2fb231f0
To: "4148" <sip:4148@customer.com>;tag=EE625669-435859C8
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: af04dc4c-fc6ea483-38848532@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a8e32c8
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7ac82345
To: "Office 106" <sip:4120@customer.com>;tag=3F364E51-5B222F0
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 9f66077-d16d29c6-d057424d@172.30.0.180
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54c8630b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as4f80e97d
To: "4136" <sip:4136@customer.com>;tag=EBBF8BD9-54CCB672
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: bef4b4f0-29e154c9-65f17362@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2a10e6db
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as15700ff6
To: "Leon Zhong" <sip:4134@customer.com>;tag=32A0D36-967AF257
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 7cdaf61-6dca67e6-6d8a2347@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Really destroying SIP dialog '55f60a00-130f7c9b-436d40b6@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK466d36fc
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as17c07070
To: "4142" <sip:4142@172.30.0.198:5060>;tag=6758A022-531979ED
CSeq: 102 OPTIONS
Call-ID: 3edf56554495cb1378d354767929739b@172.30.0.10:5060
Contact: <sip:4142@172.30.0.198:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (14 headers 0 lines) ---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62d1721a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as73c59a16
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5f506dd6493852d0721abef64f117c08@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKa9762304B841893F
From: "4168" <sip:4168@customer.com>;tag=530ADCB5-91794658
To: <sip:4148@customer.com>;tag=as5cff5298
CSeq: 4 SUBSCRIBE
Call-ID: 59688f55-c01b9f8-f830e953@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4168", realm="asterisk", nonce="5a9e28b5", uri="sip:4148@172.30.0.10:5060", response="56fa07a7da149b140f4fe23b984ec49d", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (16 headers 0 lines) ---
[2011-11-03 06:46:02] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:02] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:46:02] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1268fa28
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6ae6471f
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 698c1e0c11b7a3e07b3a01166134be89@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
burnabyasterisk*CLI>
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.197:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7746b9ff
From: <sip:4134@customer.com>;tag=as77292756
To: "4115" <sip:4115@customer.com>;tag=FA968761-5726767C
CSeq: 103 NOTIFY
Call-ID: 4844fd69-2c125084-133a351f@172.30.0.197
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (10 headers 0 lines) ---
[2011-11-03 06:46:02] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:02]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.197:5060
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK378ea673
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6e02a2c0
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 039134d54c922e5279d167e65f157b75@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:02] Retransmitting #5 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK66242fbd
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as16af495f
To: "4137" <sip:4137@customer.com>;tag=2E7E6F0B-853430C8
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 445d5a75-4e14aa6a-9075447b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f85076c
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as7bfbafb7
To: "Leon Zhong" <sip:4134@customer.com>;tag=A4F06B6C-7E0DE9D5
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: af31e56f-c591f05c-b8e08805@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #5 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16a8ec92
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as266d8d99
To: "empty station" <sip:4119@customer.com>;tag=46E2AC2F-7BB5AE26
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 32739794-c9520957-f6441ae@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22f715bb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08ad6527
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2e051206521ace410e55bcfd14ccabc3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:02] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4141' is now UNREACHABLE!  Last qualify: 49
[2011-11-03 06:46:02] Really destroying SIP dialog '2e051206521ace410e55bcfd14ccabc3@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:02] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK282b7d8c
Max-Forwards: 70
From: <sip:4137@customer.com>;tag=as4eef1d4c
To: "4124" <sip:4124@customer.com>;tag=D4B40ECE-879C1219
Contact: <sip:4137@172.30.0.10:5060>
Call-ID: 61543496-3b870d21-3bdb7ca4@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4137">
<address uri="sip:4137@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] set_destination: Parsing <sip:4142@172.30.0.198:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.198:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.198:5060:
NOTIFY sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e29496b
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as0ebc6145
To: "4142" <sip:4142@customer.com>;tag=74D9A65D-DC6455DC
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: f2d427b3-d153d31a-56517205@172.30.0.198
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4142@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4142
[2011-11-03 06:46:02] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04348c92
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as6a30eab9
To: "4118" <sip:4118@customer.com>;tag=BF39822B-83BD6000
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: a641b9eb-8a57e7c0-168deb55@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4118
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4147 (queued)
burnabyasterisk*CLI>
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4128 (queued)
[2011-11-03 06:46:02] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK559240b6
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4141
[2011-11-03 06:46:02] set_destination: Parsing <sip:4123@172.30.0.184:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.184:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d722e87
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as55400703
To: "4123" <sip:4123@customer.com>;tag=E9FBB280-4DEB8557
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 2ec5aa92-d2233079-7b3c7858@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4141[default-local] new state Unavailable for Notify User 4123
burnabyasterisk*CLI>
burnabyasterisk*CLI>
[2011-11-03 06:46:02] Really destroying SIP dialog '4844fd69-2c125084-133a351f@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0eb61b61
From: <sip:4118@customer.com>;tag=as7cecdea0
To: "4168" <sip:4168@customer.com>;tag=44916A20-951EF5BB
CSeq: 103 NOTIFY
Call-ID: ae31c99c-fff7b197-b71cfe0a@172.30.0.187
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (10 headers 0 lines) ---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5d84ee
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0f964613
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4bfef01f561aa11355bf50266675edc4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:02] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4168' is now UNREACHABLE!  Last qualify: 40
[2011-11-03 06:46:02] Really destroying SIP dialog '4bfef01f561aa11355bf50266675edc4@172.30.0.10:5060' Method: OPTIONS
burnabyasterisk*CLI>
[2011-11-03 06:46:02] Retransmitting #6 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK17335d8b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as5ede67d3
To: "4148" <sip:4148@customer.com>;tag=A378C3FF-D698038E
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: fa5374ee-dc9788b5-59616574@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c360bcd
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as45b9714a
To: "Office 108" <sip:4128@customer.com>;tag=BFDCB33E-CD5B5BA7
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 1f0e8cd6-f0b74f7f-c83fed80@172.30.0.182
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
[2011-11-03 06:46:02] Really destroying SIP dialog 'f4d30364-728ed7ff-47d7fe1a@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.198:5060:
NOTIFY sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e29496b
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as0ebc6145
To: "4142" <sip:4142@customer.com>;tag=74D9A65D-DC6455DC
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: f2d427b3-d153d31a-56517205@172.30.0.198
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4142@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5fb2bd9e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as79f82d48
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2f435bb9010529d15021f9d34b59a20e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:02] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4124' is now UNREACHABLE!  Last qualify: 59
[2011-11-03 06:46:02] Really destroying SIP dialog '2f435bb9010529d15021f9d34b59a20e@172.30.0.10:5060' Method: OPTIONS
burnabyasterisk*CLI>
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:02] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK695ee698
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4173
[2011-11-03 06:46:02] set_destination: Parsing <sip:4149@172.30.0.188:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.188:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36bb77a6
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as29d46f72
To: "4149" <sip:4149@customer.com>;tag=1A036AB2-15599CC1
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: c900a514-645b1ad3-24e6613a@172.30.0.188
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4149
[2011-11-03 06:46:02] set_destination: Parsing <sip:4138@172.30.0.163:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.163:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e443b0c
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as3e8cdda8
To: "4138" <sip:4138@customer.com>;tag=C87DED41-A6743BBA
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: e745cfbd-1ebc4c6-b065648b@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4138
[2011-11-03 06:46:02] set_destination: Parsing <sip:4122@172.30.0.176:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.176:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ffd7ca5
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as17a56026
To: "4122" <sip:4122@customer.com>;tag=411DC2D4-7DCDD2B
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 33aaa645-64e7904c-f13a5d23@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]   == Extension Changed 4124[default-local] new state Unavailable for Notify User 4122
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04348c92
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as6a30eab9
To: "4118" <sip:4118@customer.com>;tag=BF39822B-83BD6000
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: a641b9eb-8a57e7c0-168deb55@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
burnabyasterisk*CLI>
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.184:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c0d459b
From: <sip:4115@customer.com>;tag=as08562884
To: "4123" <sip:4123@customer.com>;tag=80722E46-D9C10ACD
CSeq: 105 NOTIFY
Call-ID: 75b14602-6af71369-71053ec8@172.30.0.184
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (10 headers 0 lines) ---
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK559240b6
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d722e87
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as55400703
To: "4123" <sip:4123@customer.com>;tag=E9FBB280-4DEB8557
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 2ec5aa92-d2233079-7b3c7858@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission fa5374ee-dc9788b5-59616574@172.30.0.192 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6522ms with no response
[2011-11-03 06:46:02] Really destroying SIP dialog '11165c14-a822c8af-942886ca@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:46:02] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21278db9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as208ce64c
To: "4145" <sip:4145@customer.com>;tag=196B7755-1C8E6946
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 9dd773e-6947e73b-b12100b4@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
[2011-11-03 06:46:02] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:46:02] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:46:02] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2024ac3b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as754e3632
To: "Office 131" <sip:4113@customer.com>;tag=B73F6B3E-E85CC275
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f02c36fd-a6d8f84-6155c2db@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ecdcc2f
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as1fff0b57
To: "4136" <sip:4136@customer.com>;tag=F37A13B7-5AA7ACD0
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 39b92ead-60712846-68a8f09f@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
[2011-11-03 06:46:02] Really destroying SIP dialog 'fa5374ee-dc9788b5-59616574@172.30.0.192' Method: SUBSCRIBE
burnabyasterisk*CLI>
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4115@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKb8c42a18C00F1CD3
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
CSeq: 3 SUBSCRIBE
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="3a54b1f4", uri="sip:4115@customer.com:5060", response="bfc977b9d0a65817564f0c5eaac223f2", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (16 headers 0 lines) ---
[2011-11-03 06:46:02] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:46:02] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55'
[2011-11-03 06:46:02]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKb8c42a18C00F1CD3;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30b415df", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:02] Scheduling destruction of SIP dialog '73da2ff0-edbeaf8b-16292a42@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:02] Retransmitting #2 (no NAT) to 172.30.0.198:5060:
NOTIFY sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e29496b
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as0ebc6145
To: "4142" <sip:4142@customer.com>;tag=74D9A65D-DC6455DC
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: f2d427b3-d153d31a-56517205@172.30.0.198
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4142@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK695ee698
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 28f73857-8867c578-b92b406d@172.30.0.181 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6373ms with no response
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36bb77a6
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as29d46f72
To: "4149" <sip:4149@customer.com>;tag=1A036AB2-15599CC1
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: c900a514-645b1ad3-24e6613a@172.30.0.188
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04348c92
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as6a30eab9
To: "4118" <sip:4118@customer.com>;tag=BF39822B-83BD6000
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: a641b9eb-8a57e7c0-168deb55@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e443b0c
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as3e8cdda8
To: "4138" <sip:4138@customer.com>;tag=C87DED41-A6743BBA
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: e745cfbd-1ebc4c6-b065648b@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 377b553a-764724af-434d67b0@172.30.0.163 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6372ms with no response
[2011-11-03 06:46:02] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ffd7ca5
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as17a56026
To: "4122" <sip:4122@customer.com>;tag=411DC2D4-7DCDD2B
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 33aaa645-64e7904c-f13a5d23@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK559240b6
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e45588d
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 1ebb296f-a98e3456-dd0d7c0d@172.30.0.176 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6377ms with no response
[2011-11-03 06:46:02] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d722e87
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as55400703
To: "4123" <sip:4123@customer.com>;tag=E9FBB280-4DEB8557
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 2ec5aa92-d2233079-7b3c7858@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
burnabyasterisk*CLI>
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b008ef1
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as08b6f44c
To: "4124" <sip:4124@customer.com>;tag=DAA75691-6EF42394
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: b5bf6aea-f1839b95-b22ea038@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3472001b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as2fb231f0
To: "4148" <sip:4148@customer.com>;tag=EE625669-435859C8
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: af04dc4c-fc6ea483-38848532@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a8e32c8
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7ac82345
To: "Office 106" <sip:4120@customer.com>;tag=3F364E51-5B222F0
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 9f66077-d16d29c6-d057424d@172.30.0.180
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54c8630b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as4f80e97d
To: "4136" <sip:4136@customer.com>;tag=EBBF8BD9-54CCB672
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: bef4b4f0-29e154c9-65f17362@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2a10e6db
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as15700ff6
To: "Leon Zhong" <sip:4134@customer.com>;tag=32A0D36-967AF257
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 7cdaf61-6dca67e6-6d8a2347@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:02]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4124@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKd2780ee84CE576FF
From: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
To: <sip:4124@customer.com>;tag=as75491cdd
CSeq: 3 SUBSCRIBE
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="02058cf1", uri="sip:4124@customer.com:5060", response="017e830ded133152476a8c2aeb7f1fc3", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:02] --- (16 headers 0 lines) ---
[2011-11-03 06:46:02] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:46:02] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8'
[2011-11-03 06:46:02]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKd2780ee84CE576FF;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
To: <sip:4124@customer.com>;tag=as75491cdd
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c3e8f0a", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:03] Scheduling destruction of SIP dialog 'bfbd9e39-ced80418-ec8d31af@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:03] Retransmitting #6 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK33514b95
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as6aba6b43
To: "4149" <sip:4149@customer.com>;tag=E2CE1FF8-BC9E5817
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: 7c437ced-f09b2644-c09eb483@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
burnabyasterisk*CLI>
burnabyasterisk*CLI>
burnabyasterisk*CLI>
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2024ac3b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as754e3632
To: "Office 131" <sip:4113@customer.com>;tag=B73F6B3E-E85CC275
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f02c36fd-a6d8f84-6155c2db@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #6 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35d5f8d7
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as648f305f
To: "4116" <sip:4116@customer.com>;tag=503C3B5D-4EDC272E
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: a4528d90-6cfe64a9-69eac99a@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #6 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01c30cf7
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as0d4ac688
To: "4145" <sip:4145@customer.com>;tag=206B3278-C04B246D
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: a4ed9564-8af89fe9-41db734a@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.197:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2914f90f
From: <sip:4116@customer.com>;tag=as4437c966
To: "4115" <sip:4115@customer.com>;tag=A9950082-67964C1D
CSeq: 103 NOTIFY
Call-ID: e596c46f-a697628a-13dc1225@172.30.0.197
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:03] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 7c437ced-f09b2644-c09eb483@172.30.0.188 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6435ms with no response
[2011-11-03 06:46:03] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a4528d90-6cfe64a9-69eac99a@172.30.0.199 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK695ee698
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36bb77a6
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as29d46f72
To: "4149" <sip:4149@customer.com>;tag=1A036AB2-15599CC1
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: c900a514-645b1ad3-24e6613a@172.30.0.188
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e443b0c
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as3e8cdda8
To: "4138" <sip:4138@customer.com>;tag=C87DED41-A6743BBA
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: e745cfbd-1ebc4c6-b065648b@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ffd7ca5
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as17a56026
To: "4122" <sip:4122@customer.com>;tag=411DC2D4-7DCDD2B
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 33aaa645-64e7904c-f13a5d23@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4140@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK3ad677a0FA3CBF3B
From: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
To: <sip:4140@customer.com>;tag=as73332664
CSeq: 4 SUBSCRIBE
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4140", realm="asterisk", nonce="57c05799", uri="sip:4140@172.30.0.10:5060", response="93a52b0adee927165a99d813a6d084ff", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
burnabyasterisk*CLI>
[2011-11-03 06:46:03] --- (16 headers 0 lines) ---
[2011-11-03 06:46:03] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:46:03] Looking for 4140 in default-local (domain customer.com:5060)
[2011-11-03 06:46:03]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK3ad677a0FA3CBF3B;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
To: <sip:4140@customer.com>;tag=as73332664
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:03] set_destination: Parsing <sip:4140@172.30.0.175:5060> for address/port to send to
[2011-11-03 06:46:03] set_destination: set destination to 172.30.0.175:5060
[2011-11-03 06:46:03] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a4ed9564-8af89fe9-41db734a@172.30.0.167 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6445ms with no response
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2024ac3b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as754e3632
To: "Office 131" <sip:4113@customer.com>;tag=B73F6B3E-E85CC275
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f02c36fd-a6d8f84-6155c2db@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.177:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d5151e2
From: <sip:4137@customer.com>;tag=as0fbc87a7
To: "4135" <sip:4135@customer.com>;tag=E95C8A73-A146210
CSeq: 104 NOTIFY
Call-ID: 6f97a0da-f3f2d62b-9fe9d168@172.30.0.177
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:03] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:03]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.177:5060
[2011-11-03 06:46:03] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62d1721a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as73c59a16
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5f506dd6493852d0721abef64f117c08@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:03] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4122' is now UNREACHABLE!  Last qualify: 84
[2011-11-03 06:46:03] Really destroying SIP dialog '5f506dd6493852d0721abef64f117c08@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4116 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:46:03] set_destination: Parsing <sip:4149@172.30.0.188:5060> for address/port to send to
[2011-11-03 06:46:03] set_destination: set destination to 172.30.0.188:5060
[2011-11-03 06:46:03] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4149
[2011-11-03 06:46:03] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:46:03] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:46:03] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK106817c6
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as58f94645
To: "4148" <sip:4148@customer.com>;tag=A421AA73-D0CB46A2
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8308066-263455ed-e9eab26c@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]   == Extension Changed 4122[default-local] new state Unavailable for Notify User 4148
[2011-11-03 06:46:03] Really destroying SIP dialog '6f97a0da-f3f2d62b-9fe9d168@172.30.0.177' Method: SUBSCRIBE
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.163:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a5bc1df
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4d549692
To: <sip:4138@172.30.0.163:5060>;tag=8BFD62CE-BDC5C973
CSeq: 102 OPTIONS
Call-ID: 470d162651401a3319824e346a93bfc4@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (9 headers 0 lines) ---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1268fa28
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6ae6471f
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 698c1e0c11b7a3e07b3a01166134be89@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.198:5060:
NOTIFY sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e29496b
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as0ebc6145
To: "4142" <sip:4142@customer.com>;tag=74D9A65D-DC6455DC
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: f2d427b3-d153d31a-56517205@172.30.0.198
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4142@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04348c92
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as6a30eab9
To: "4118" <sip:4118@customer.com>;tag=BF39822B-83BD6000
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: a641b9eb-8a57e7c0-168deb55@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK559240b6
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d722e87
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as55400703
To: "4123" <sip:4123@customer.com>;tag=E9FBB280-4DEB8557
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 2ec5aa92-d2233079-7b3c7858@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #6 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4edcbecf
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.179:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ac0e8a4
From: <sip:4131@customer.com>;tag=as5f7e32a1
To: "Leon Zhong" <sip:4134@customer.com>;tag=E3D41D55-2ACD6BCA
CSeq: 104 NOTIFY
Call-ID: 985c71c5-c904837a-60f82c4b@172.30.0.179
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:03] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:03]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.179:5060
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK106817c6
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as58f94645
To: "4148" <sip:4148@customer.com>;tag=A421AA73-D0CB46A2
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8308066-263455ed-e9eab26c@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5438a269
From: <sip:4130@customer.com>;tag=as70865b2e
To: "Office 131" <sip:4113@customer.com>;tag=16D6ACA6-61364B5D
CSeq: 103 NOTIFY
Call-ID: 2b297ea3-c6eac54a-1a74dcc1@172.30.0.195
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:03] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:03]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.195:5060
[2011-11-03 06:46:03] Retransmitting #6 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7dfdb1da
Max-Forwards: 70
From: <sip:4137@customer.com>;tag=as790d26be
To: "4140" <sip:4140@customer.com>;tag=C3F351A2-A3B1810D
Contact: <sip:4137@172.30.0.10:5060>
Call-ID: ed08d2df-9f1b6092-9f09697d@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4137">
<address uri="sip:4137@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.164:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5fb2bd9e
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as79f82d48
To: <sip:4124@172.30.0.164:5060>;tag=3078579D-78D31B80
CSeq: 102 OPTIONS
Call-ID: 2f435bb9010529d15021f9d34b59a20e@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (9 headers 0 lines) ---
[2011-11-03 06:46:03] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK378ea673
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6e02a2c0
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 039134d54c922e5279d167e65f157b75@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:45:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:03] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4140' is now UNREACHABLE!  Last qualify: 75
[2011-11-03 06:46:03] Really destroying SIP dialog '039134d54c922e5279d167e65f157b75@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:46:03] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:46:03] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:46:03] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a1a0149
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as5881d52c
To: "Office 106" <sip:4120@customer.com>;tag=D07B081B-73C9538A
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.180
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4120
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:46:03] set_destination: Parsing <sip:4123@172.30.0.184:5060> for address/port to send to
[2011-11-03 06:46:03] set_destination: set destination to 172.30.0.184:5060
[2011-11-03 06:46:03] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4123
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:46:03] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:46:03] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:46:03] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0c35e750
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as54deba17
To: "4116" <sip:4116@customer.com>;tag=39B600A4-7C17D79D
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a99aa77a-c3145783-cd4773e4@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4116
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4125 (queued)
[2011-11-03 06:46:03]   == Extension Changed 4140[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4108@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK25ee265819E36C31
From: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
To: <sip:4108@customer.com>;tag=as1105be02
CSeq: 3 SUBSCRIBE
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4128", realm="asterisk", nonce="51b53ce4", uri="sip:4108@customer.com:5060", response="37843f022d235ece23296e46b46577e3", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (16 headers 0 lines) ---
[2011-11-03 06:46:03] Found peer '4128' for '4128' from 172.30.0.182:5060
[2011-11-03 06:46:03] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56'
[2011-11-03 06:46:03]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK25ee265819E36C31;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
To: <sip:4108@customer.com>;tag=as1105be02
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4806cf34", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:03] Scheduling destruction of SIP dialog '4d5c1f87-46dbba48-db35e6a1@172.30.0.182' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK695ee698
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36bb77a6
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as29d46f72
To: "4149" <sip:4149@customer.com>;tag=1A036AB2-15599CC1
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: c900a514-645b1ad3-24e6613a@172.30.0.188
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e443b0c
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as3e8cdda8
To: "4138" <sip:4138@customer.com>;tag=C87DED41-A6743BBA
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: e745cfbd-1ebc4c6-b065648b@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ffd7ca5
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as17a56026
To: "4122" <sip:4122@customer.com>;tag=411DC2D4-7DCDD2B
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 33aaa645-64e7904c-f13a5d23@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ed08d2df-9f1b6092-9f09697d@172.30.0.175 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6437ms with no response
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f8450d3
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as089c2896
To: <sip:4119@172.30.0.191:5060>;tag=D49436AF-5C64BEA6
CSeq: 102 OPTIONS
Call-ID: 25d817a07a30bca302a056aa27ae1394@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (9 headers 0 lines) ---
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK106817c6
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as58f94645
To: "4148" <sip:4148@customer.com>;tag=A421AA73-D0CB46A2
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8308066-263455ed-e9eab26c@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0c35e750
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as54deba17
To: "4116" <sip:4116@customer.com>;tag=39B600A4-7C17D79D
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a99aa77a-c3145783-cd4773e4@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Really destroying SIP dialog 'ae31c99c-fff7b197-b71cfe0a@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2024ac3b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as754e3632
To: "Office 131" <sip:4113@customer.com>;tag=B73F6B3E-E85CC275
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f02c36fd-a6d8f84-6155c2db@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.192:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4765daad
From: <sip:4116@customer.com>;tag=as5a619690
To: "4148" <sip:4148@customer.com>;tag=50FDB04E-A5671715
CSeq: 106 NOTIFY
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.192
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:03] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:03]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.192:5060
[2011-11-03 06:46:03] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62bd35b3
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as5ed1c0df
To: "4147" <sip:4147@customer.com>;tag=CB4DDEC5-AA08E856
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: c318f314-94094c8d-674f3cde@172.30.0.183
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.170:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2eafe888
From: <sip:4118@customer.com>;tag=as170b1f9f
To: "Office 105" <sip:4131@customer.com>;tag=76D48057-204F6ACA
CSeq: 103 NOTIFY
Call-ID: 10ed4427-bdd931a-19963f45@172.30.0.170
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:03] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c318f314-94094c8d-674f3cde@172.30.0.183 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6416ms with no response
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a1a0149
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as5881d52c
To: "Office 106" <sip:4120@customer.com>;tag=D07B081B-73C9538A
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.180
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4173@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKe1e9720A84419BB
From: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
To: <sip:4173@customer.com>;tag=as63503940
CSeq: 3 SUBSCRIBE
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4108", realm="asterisk", nonce="09e68c16", uri="sip:4173@customer.com:5060", response="d2f64a78c10bd52d18ecb50e23429ebe", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (16 headers 0 lines) ---
[2011-11-03 06:46:03] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:46:03] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9'
[2011-11-03 06:46:03]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKe1e9720A84419BB;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
To: <sip:4173@customer.com>;tag=as63503940
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="002ff46a", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:03] Scheduling destruction of SIP dialog 'bc846304-17b3879f-6fd7fdba@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0c35e750
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as54deba17
To: "4116" <sip:4116@customer.com>;tag=39B600A4-7C17D79D
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a99aa77a-c3145783-cd4773e4@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4106@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK2374861bAB94962E
From: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
To: <sip:4106@customer.com>;tag=as616b4d7c
CSeq: 4 SUBSCRIBE
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4124", realm="asterisk", nonce="22e7efa1", uri="sip:4106@172.30.0.10:5060", response="b6dbab6dca14bda6926c458d0bd02878", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (16 headers 0 lines) ---
[2011-11-03 06:46:03] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:46:03] Looking for 4106 in default-local (domain customer.com:5060)
[2011-11-03 06:46:03]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK2374861bAB94962E;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
To: <sip:4106@customer.com>;tag=as616b4d7c
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:03] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:46:03] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:46:03] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45edadf2
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as616b4d7c
To: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK266230cf
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as7c6d37d6
To: "4147" <sip:4147@customer.com>;tag=4C1D584F-2F9D61D0
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 3af79169-77fdc85a-8cf41d63@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK610ec4e0
From: <sip:4116@customer.com>;tag=as3b261357
To: "Office 113" <sip:4171@customer.com>;tag=DE9E2A48-9DE3D0C3
CSeq: 105 NOTIFY
Call-ID: e69ec24d-a1b6e62c-e18efe77@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:03] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:03]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.165:5060
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.176:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62d1721a
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as73c59a16
To: <sip:4122@172.30.0.176:5060>;tag=E829B4C3-A84E716A
CSeq: 102 OPTIONS
Call-ID: 5f506dd6493852d0721abef64f117c08@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (9 headers 0 lines) ---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK106817c6
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as58f94645
To: "4148" <sip:4148@customer.com>;tag=A421AA73-D0CB46A2
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8308066-263455ed-e9eab26c@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45edadf2
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as616b4d7c
To: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:03]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5438a269
From: <sip:4130@customer.com>;tag=as70865b2e
To: "Office 131" <sip:4113@customer.com>;tag=16D6ACA6-61364B5D
CSeq: 103 NOTIFY
Call-ID: 2b297ea3-c6eac54a-1a74dcc1@172.30.0.195
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:03] --- (10 headers 0 lines) ---
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4135@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK427a3bf8EE1CB8D1
From: "Office 108" <sip:4128@customer.com>;tag=2B68B9DA-3214F2E3
To: <sip:4135@customer.com>;tag=as4e5800f1
CSeq: 4 SUBSCRIBE
Call-ID: 46c4bc42-3be57b0b-c015a82c@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4128", realm="asterisk", nonce="06169684", uri="sip:4135@172.30.0.10:5060", response="845c01c75e70ddca8a79645841d14462", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (16 headers 0 lines) ---
[2011-11-03 06:46:04] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:04]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK427a3bf8EE1CB8D1;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=2B68B9DA-3214F2E3
To: <sip:4135@customer.com>;tag=as4e5800f1
Call-ID: 46c4bc42-3be57b0b-c015a82c@172.30.0.182
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:04] Retransmitting #6 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48ce56c6
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1a438c02
To: "4145" <sip:4145@customer.com>;tag=E42AA443-EC87CB1C
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: bc992562-71188b8f-79764b78@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.198:5060:
NOTIFY sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e29496b
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as0ebc6145
To: "4142" <sip:4142@customer.com>;tag=74D9A65D-DC6455DC
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: f2d427b3-d153d31a-56517205@172.30.0.198
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4142@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Really destroying SIP dialog '46c4bc42-3be57b0b-c015a82c@172.30.0.182' Method: SUBSCRIBE
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK761bc3f5
From: <sip:4171@customer.com>;tag=as2ddd8afa
To: "4136" <sip:4136@customer.com>;tag=D6E3A34-13B5BD0D
CSeq: 106 NOTIFY
Call-ID: a032fe18-d29dbf1-23fab88a@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (11 headers 0 lines) ---
[2011-11-03 06:46:04] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04348c92
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as6a30eab9
To: "4118" <sip:4118@customer.com>;tag=BF39822B-83BD6000
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: a641b9eb-8a57e7c0-168deb55@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK559240b6
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d722e87
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as55400703
To: "4123" <sip:4123@customer.com>;tag=E9FBB280-4DEB8557
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 2ec5aa92-d2233079-7b3c7858@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5d84ee
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0f964613
To: <sip:4168@172.30.0.187:5060>;tag=4E1AB3C0-6AB0D85B
CSeq: 102 OPTIONS
Call-ID: 4bfef01f561aa11355bf50266675edc4@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (9 headers 0 lines) ---
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission bc992562-71188b8f-79764b78@172.30.0.167 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6394ms with no response
[2011-11-03 06:46:04] Retransmitting #6 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK51c7d887
Max-Forwards: 70
From: <sip:4131@customer.com>;tag=as7ab1ecb2
To: "4123" <sip:4123@customer.com>;tag=694DEA52-BFBD9E39
Contact: <sip:4131@172.30.0.10:5060>
Call-ID: 889021c4-847ffbbb-5bbf2c2a@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4131">
<address uri="sip:4131@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #6 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72fa3857
Max-Forwards: 70
From: <sip:4131@customer.com>;tag=as5ec998b0
To: "4140" <sip:4140@customer.com>;tag=2FDAE57-E88448CA
Contact: <sip:4131@172.30.0.10:5060>
Call-ID: c3aff354-abfd920f-63da7342@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4131">
<address uri="sip:4131@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21278db9
From: <sip:4136@customer.com>;tag=as208ce64c
To: "4145" <sip:4145@customer.com>;tag=196B7755-1C8E6946
CSeq: 103 NOTIFY
Call-ID: 9dd773e-6947e73b-b12100b4@172.30.0.167
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (10 headers 0 lines) ---
[2011-11-03 06:46:04] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:04]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.167:5060
[2011-11-03 06:46:04] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK542f3557
Max-Forwards: 70
From: <sip:4131@customer.com>;tag=as48965426
To: "4124" <sip:4124@customer.com>;tag=E88448CA-4AB62C75
Contact: <sip:4131@172.30.0.10:5060>
Call-ID: dc2b1046-78db9051-c3aff354@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4131">
<address uri="sip:4131@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0c35e750
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as54deba17
To: "4116" <sip:4116@customer.com>;tag=39B600A4-7C17D79D
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a99aa77a-c3145783-cd4773e4@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45edadf2
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as616b4d7c
To: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9ffac873-472d4810-28b1a2a9@172.30.0.168 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6280ms with no response
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4961221f
From: <sip:4130@customer.com>;tag=as76601e10
To: "Office 107" <sip:4141@customer.com>;tag=F07BC7E6-6E9E20F3
CSeq: 105 NOTIFY
Call-ID: 28f73857-8867c578-b92b406d@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (10 headers 0 lines) ---
[2011-11-03 06:46:04] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:46:04] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:46:04] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission dc2b1046-78db9051-c3aff354@172.30.0.164 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6388ms with no response
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.195:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKcd187b907D65CF27
From: "Office 131" <sip:4113@customer.com>;tag=6632B9A1-EBB6AE8
To: <sip:4113@customer.com>
CSeq: 3 REGISTER
Call-ID: b039439d-79868a24-bf6a4b7b@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4113", realm="asterisk", nonce="08d61a6f", uri="sip:customer.com:5060", response="f730f07b09fe5a0805ad5a272dd081ed", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (13 headers 0 lines) ---
[2011-11-03 06:46:04] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:46:04]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKcd187b907D65CF27;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=6632B9A1-EBB6AE8
To: <sip:4113@customer.com>;tag=as780a9465
Call-ID: b039439d-79868a24-bf6a4b7b@172.30.0.195
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c3bc11c"
Content-Length: 0


<------------>
[2011-11-03 06:46:04] Scheduling destruction of SIP dialog 'b039439d-79868a24-bf6a4b7b@172.30.0.195' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 889021c4-847ffbbb-5bbf2c2a@172.30.0.184 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6540ms with no response
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c360bcd
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as45b9714a
To: "Office 108" <sip:4128@customer.com>;tag=BFDCB33E-CD5B5BA7
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 1f0e8cd6-f0b74f7f-c83fed80@172.30.0.182
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c3aff354-abfd920f-63da7342@172.30.0.175 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6540ms with no response
[2011-11-03 06:46:04] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a1a0149
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as5881d52c
To: "Office 106" <sip:4120@customer.com>;tag=D07B081B-73C9538A
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.180
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1268fa28
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6ae6471f
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 698c1e0c11b7a3e07b3a01166134be89@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:04] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4115' is now UNREACHABLE!  Last qualify: 38
[2011-11-03 06:46:04] Really destroying SIP dialog '698c1e0c11b7a3e07b3a01166134be89@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:46:04] set_destination: Parsing <sip:4140@172.30.0.175:5060> for address/port to send to
[2011-11-03 06:46:04] set_destination: set destination to 172.30.0.175:5060
[2011-11-03 06:46:04] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6bf6199c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4140
[2011-11-03 06:46:04] set_destination: Parsing <sip:4168@172.30.0.187:5060> for address/port to send to
[2011-11-03 06:46:04] set_destination: set destination to 172.30.0.187:5060
[2011-11-03 06:46:04] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1acf8c29
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as58f1accb
To: "4168" <sip:4168@customer.com>;tag=93C331FA-B8037925
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4168
[2011-11-03 06:46:04] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:46:04] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:46:04] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK385dbd1c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as01d85198
To: "Leon Zhong" <sip:4134@customer.com>;tag=1C6DE198-7CE57F91
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 857aaf26-f301a987-4ad40894@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4134
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:46:04] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:46:04] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:46:04] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK33510b01
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as656896f7
To: "4148" <sip:4148@customer.com>;tag=13B5A662-4E562EC9
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: b1905049-17fabaa8-f17cb0bf@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4148
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK695ee698
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36bb77a6
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as29d46f72
To: "4149" <sip:4149@customer.com>;tag=1A036AB2-15599CC1
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: c900a514-645b1ad3-24e6613a@172.30.0.188
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e443b0c
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as3e8cdda8
To: "4138" <sip:4138@customer.com>;tag=C87DED41-A6743BBA
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: e745cfbd-1ebc4c6-b065648b@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ffd7ca5
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as17a56026
To: "4122" <sip:4122@customer.com>;tag=411DC2D4-7DCDD2B
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 33aaa645-64e7904c-f13a5d23@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e21e2e85-937ffcb6-724acd53@172.30.0.167 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6412ms with no response
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9a158161-86e251e6-45d64547@172.30.0.177 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6406ms with no response
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 573873a8-23107a83-5ca380d6@172.30.0.170 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6405ms with no response
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ac2c1e2b-e40ea252-1c273f49@172.30.0.195 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6402ms with no response
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e69ec24d-a1b6e62c-e18efe77@172.30.0.165 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2024ac3b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as754e3632
To: "Office 131" <sip:4113@customer.com>;tag=B73F6B3E-E85CC275
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f02c36fd-a6d8f84-6155c2db@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #6 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6b91b9db
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as448c7fab
To: "4123" <sip:4123@customer.com>;tag=D46D9FA5-5496FFE4
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: d0cb46a2-dee0a109-24910168@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Really destroying SIP dialog 'e69ec24d-a1b6e62c-e18efe77@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.197:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKcb796b902FAA662B
From: "4115" <sip:4115@customer.com>;tag=FFA93B0D-23BE1028
To: <sip:4115@customer.com>
CSeq: 3 REGISTER
Call-ID: 5a7cf879-53c09394-e0e2c02f@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4115", realm="asterisk", nonce="778c7497", uri="sip:customer.com:5060", response="bb0ebe9ac3c5f3f54c1e1f9d87a4c6d2", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (13 headers 0 lines) ---
[2011-11-03 06:46:04] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:46:04]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKcb796b902FAA662B;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=FFA93B0D-23BE1028
To: <sip:4115@customer.com>;tag=as47295813
Call-ID: 5a7cf879-53c09394-e0e2c02f@172.30.0.197
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7bcb5b6b"
Content-Length: 0


<------------>
[2011-11-03 06:46:04] Scheduling destruction of SIP dialog '5a7cf879-53c09394-e0e2c02f@172.30.0.197' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK755a9481
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission d0cb46a2-dee0a109-24910168@172.30.0.184 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6483ms with no response
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e45588d
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b008ef1
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as08b6f44c
To: "4124" <sip:4124@customer.com>;tag=DAA75691-6EF42394
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: b5bf6aea-f1839b95-b22ea038@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3472001b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as2fb231f0
To: "4148" <sip:4148@customer.com>;tag=EE625669-435859C8
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: af04dc4c-fc6ea483-38848532@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a8e32c8
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7ac82345
To: "Office 106" <sip:4120@customer.com>;tag=3F364E51-5B222F0
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 9f66077-d16d29c6-d057424d@172.30.0.180
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54c8630b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as4f80e97d
To: "4136" <sip:4136@customer.com>;tag=EBBF8BD9-54CCB672
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: bef4b4f0-29e154c9-65f17362@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2a10e6db
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as15700ff6
To: "Leon Zhong" <sip:4134@customer.com>;tag=32A0D36-967AF257
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 7cdaf61-6dca67e6-6d8a2347@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6bf6199c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1acf8c29
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as58f1accb
To: "4168" <sip:4168@customer.com>;tag=93C331FA-B8037925
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK385dbd1c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as01d85198
To: "Leon Zhong" <sip:4134@customer.com>;tag=1C6DE198-7CE57F91
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 857aaf26-f301a987-4ad40894@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK33510b01
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as656896f7
To: "4148" <sip:4148@customer.com>;tag=13B5A662-4E562EC9
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: b1905049-17fabaa8-f17cb0bf@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKbb06200e1FE8C1A9
From: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
To: <sip:4148@customer.com>;tag=as314b084b
CSeq: 3 SUBSCRIBE
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="19396a23", uri="sip:4148@customer.com:5060", response="2fb1c9abe7abd4f85136d39949ec4f3b", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (16 headers 0 lines) ---
[2011-11-03 06:46:04] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:46:04] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00'
[2011-11-03 06:46:04]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKbb06200e1FE8C1A9;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
To: <sip:4148@customer.com>;tag=as314b084b
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79135535", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:04] Scheduling destruction of SIP dialog 'b06505b4-1b81c24f-6b7fd06a@172.30.0.197' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:04] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45edadf2
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as616b4d7c
To: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff6cbdd
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as284d87e4
To: "4118" <sip:4118@customer.com>;tag=E9AFD17D-87EBC8D2
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 28129ff1-31250646-5c60a25b@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21278db9
From: <sip:4136@customer.com>;tag=as208ce64c
To: "4145" <sip:4145@customer.com>;tag=196B7755-1C8E6946
CSeq: 103 NOTIFY
Call-ID: 9dd773e-6947e73b-b12100b4@172.30.0.167
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (10 headers 0 lines) ---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4127@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKfd45eccb71164B88
From: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
To: <sip:4127@customer.com>;tag=as1df1c6ea
CSeq: 5 SUBSCRIBE
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="358f01ff", uri="sip:4127@172.30.0.10:5060", response="fde67921165c19877b6751f146a0bd9e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (16 headers 0 lines) ---
[2011-11-03 06:46:04] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:46:04] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280'
[2011-11-03 06:46:04]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKfd45eccb71164B88;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
To: <sip:4127@customer.com>;tag=as1df1c6ea
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 5 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6622e16f", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:04] Scheduling destruction of SIP dialog 'b0ba94bb-b41a10b8-a3494831@172.30.0.177' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:04] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6bf6199c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1acf8c29
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as58f1accb
To: "4168" <sip:4168@customer.com>;tag=93C331FA-B8037925
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK385dbd1c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as01d85198
To: "Leon Zhong" <sip:4134@customer.com>;tag=1C6DE198-7CE57F91
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 857aaf26-f301a987-4ad40894@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK33510b01
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as656896f7
To: "4148" <sip:4148@customer.com>;tag=13B5A662-4E562EC9
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: b1905049-17fabaa8-f17cb0bf@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK106817c6
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as58f94645
To: "4148" <sip:4148@customer.com>;tag=A421AA73-D0CB46A2
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8308066-263455ed-e9eab26c@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK3e2c1cdb60DC9EB0
From: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
To: <sip:4148@customer.com>;tag=as59b14bbd
CSeq: 3 SUBSCRIBE
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="037400c3", uri="sip:4148@customer.com:5060", response="a5a56c759e9c73505c6cce6178d8caf5", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (16 headers 0 lines) ---
[2011-11-03 06:46:04] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:46:04] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6'
[2011-11-03 06:46:04]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK3e2c1cdb60DC9EB0;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
To: <sip:4148@customer.com>;tag=as59b14bbd
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="217e30fe", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:04] Scheduling destruction of SIP dialog '250d68f8-3904e68d-145089e2@172.30.0.173' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:04] Really destroying SIP dialog '2b297ea3-c6eac54a-1a74dcc1@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:46:04]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4136@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK6b9bf494C4169B61
From: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
To: <sip:4136@customer.com>;tag=as1938f6a1
CSeq: 3 SUBSCRIBE
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="49daba91", uri="sip:4136@customer.com:5060", response="9d111d041b3cf2c1b460d87289af6670", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:04] --- (16 headers 0 lines) ---
[2011-11-03 06:46:04] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:46:04] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4138" <sip:4138@customer.com>;tag=742E799D-D388B26'
[2011-11-03 06:46:04]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK6b9bf494C4169B61;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
To: <sip:4136@customer.com>;tag=as1938f6a1
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ba40ead", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:04] Scheduling destruction of SIP dialog 'fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ef0ee656-7679775d-5bbfcb5c@172.30.0.192 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6416ms with no response
[2011-11-03 06:46:04] Retransmitting #6 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK11ba61a3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1c439bdf
To: "Office 108" <sip:4128@customer.com>;tag=8B0D8457-86272A98
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 9355ea5-22367536-a032d0df@172.30.0.182
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0c35e750
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as54deba17
To: "4116" <sip:4116@customer.com>;tag=39B600A4-7C17D79D
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a99aa77a-c3145783-cd4773e4@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:04] Retransmitting #7 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK349b135e
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as29c5093b
To: "4136" <sip:4136@customer.com>;tag=AF9D5AC3-4C80CDC
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: 697be950-161e5129-5d1a37c2@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Really destroying SIP dialog 'ef0ee656-7679775d-5bbfcb5c@172.30.0.192' Method: SUBSCRIBE
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.197:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1268fa28
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6ae6471f
To: <sip:4115@172.30.0.197:5060>;tag=7BB87CAC-65EA5547
CSeq: 102 OPTIONS
Call-ID: 698c1e0c11b7a3e07b3a01166134be89@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (9 headers 0 lines) ---
[2011-11-03 06:46:05] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 59688f55-c01b9f8-f830e953@172.30.0.187 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6503ms with no response
[2011-11-03 06:46:05] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK183c55e2
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6bf6199c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1acf8c29
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as58f1accb
To: "4168" <sip:4168@customer.com>;tag=93C331FA-B8037925
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK385dbd1c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as01d85198
To: "Leon Zhong" <sip:4134@customer.com>;tag=1C6DE198-7CE57F91
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 857aaf26-f301a987-4ad40894@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK33510b01
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as656896f7
To: "4148" <sip:4148@customer.com>;tag=13B5A662-4E562EC9
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: b1905049-17fabaa8-f17cb0bf@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Really destroying SIP dialog '59688f55-c01b9f8-f830e953@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04457a9c
From: <sip:4148@customer.com>;tag=as5cff5298
To: "4168" <sip:4168@customer.com>;tag=530ADCB5-91794658
CSeq: 104 NOTIFY
Call-ID: 59688f55-c01b9f8-f830e953@172.30.0.187
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (10 headers 0 lines) ---
[2011-11-03 06:46:05] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a1a0149
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as5881d52c
To: "Office 106" <sip:4120@customer.com>;tag=D07B081B-73C9538A
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.180
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14f4564b
Max-Forwards: 70
From: <sip:4131@customer.com>;tag=as1343b7a2
To: "4136" <sip:4136@customer.com>;tag=F58CF41-E982E7DA
Contact: <sip:4131@172.30.0.10:5060>
Call-ID: d67fc766-e09267bf-51c6e6d8@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4131">
<address uri="sip:4131@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6b6ef391
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as09594776
To: <sip:4113@172.30.0.195:5060>;tag=BFE4F934-28BC358B
CSeq: 102 OPTIONS
Call-ID: 0918b58e25147a4b532124bd15873178@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (9 headers 0 lines) ---
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45edadf2
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as616b4d7c
To: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21278db9
From: <sip:4136@customer.com>;tag=as208ce64c
To: "4145" <sip:4145@customer.com>;tag=196B7755-1C8E6946
CSeq: 103 NOTIFY
Call-ID: 9dd773e-6947e73b-b12100b4@172.30.0.167
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (10 headers 0 lines) ---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4145@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK92105744F9C4FD3D
From: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
To: <sip:4145@customer.com>;tag=as5f5dbb11
CSeq: 3 SUBSCRIBE
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="463f6ffd", uri="sip:4145@customer.com:5060", response="4e671ed48dc3c7a33000da4c8dba1ce2", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (16 headers 0 lines) ---
[2011-11-03 06:46:05] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:46:05] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B'
[2011-11-03 06:46:05]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK92105744F9C4FD3D;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
To: <sip:4145@customer.com>;tag=as5f5dbb11
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7b40474c", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:05] Scheduling destruction of SIP dialog '932bc7b7-ad540ef8-83c8d3d1@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22f715bb
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08ad6527
To: <sip:4141@172.30.0.181:5060>;tag=D47DD0A6-AA9248B3
CSeq: 102 OPTIONS
Call-ID: 2e051206521ace410e55bcfd14ccabc3@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (9 headers 0 lines) ---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5d84ee
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0f964613
To: <sip:4168@172.30.0.187:5060>;tag=89CC326E-84D05EB9
CSeq: 102 OPTIONS
Call-ID: 4bfef01f561aa11355bf50266675edc4@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (9 headers 0 lines) ---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.197:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7746b9ff
From: <sip:4134@customer.com>;tag=as77292756
To: "4115" <sip:4115@customer.com>;tag=FA968761-5726767C
CSeq: 103 NOTIFY
Call-ID: 4844fd69-2c125084-133a351f@172.30.0.197
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (10 headers 0 lines) ---
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0975328c
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as728e16e0
To: "4136" <sip:4136@customer.com>;tag=19F02E68-92FA0841
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: cc56bb5f-1e5b7278-c0f81851@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bKa45960e4340ACC71
From: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
To: <sip:4148@customer.com>;tag=as3e7e7001
CSeq: 4 SUBSCRIBE
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="28bd45ff", uri="sip:4148@172.30.0.10:5060", response="31181b411549596a2f46718879cb647c", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (16 headers 0 lines) ---
[2011-11-03 06:46:05] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:46:05] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:46:05]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bKa45960e4340ACC71;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
To: <sip:4148@customer.com>;tag=as3e7e7001
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:05] set_destination: Parsing <sip:4138@172.30.0.163:5060> for address/port to send to
[2011-11-03 06:46:05] set_destination: set destination to 172.30.0.163:5060
[2011-11-03 06:46:05] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ca703e2
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as3e7e7001
To: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5c031c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 445d5a75-4e14aa6a-9075447b@172.30.0.166 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6337ms with no response
[2011-11-03 06:46:05] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f85076c
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as7bfbafb7
To: "Leon Zhong" <sip:4134@customer.com>;tag=A4F06B6C-7E0DE9D5
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: af31e56f-c591f05c-b8e08805@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #6 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16a8ec92
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as266d8d99
To: "empty station" <sip:4119@customer.com>;tag=46E2AC2F-7BB5AE26
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 32739794-c9520957-f6441ae@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.198:5060:
NOTIFY sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e29496b
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as0ebc6145
To: "4142" <sip:4142@customer.com>;tag=74D9A65D-DC6455DC
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: f2d427b3-d153d31a-56517205@172.30.0.198
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4142@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f8450d3
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as089c2896
To: <sip:4119@172.30.0.191:5060>;tag=B0BB4423-5327CFCA
CSeq: 102 OPTIONS
Call-ID: 25d817a07a30bca302a056aa27ae1394@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (9 headers 0 lines) ---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04348c92
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as6a30eab9
To: "4118" <sip:4118@customer.com>;tag=BF39822B-83BD6000
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: a641b9eb-8a57e7c0-168deb55@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK559240b6
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d722e87
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as55400703
To: "4123" <sip:4123@customer.com>;tag=E9FBB280-4DEB8557
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 2ec5aa92-d2233079-7b3c7858@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK282b7d8c
Max-Forwards: 70
From: <sip:4137@customer.com>;tag=as4eef1d4c
To: "4124" <sip:4124@customer.com>;tag=D4B40ECE-879C1219
Contact: <sip:4137@172.30.0.10:5060>
Call-ID: 61543496-3b870d21-3bdb7ca4@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4137">
<address uri="sip:4137@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission af31e56f-c591f05c-b8e08805@172.30.0.179 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6398ms with no response
[2011-11-03 06:46:05] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 32739794-c9520957-f6441ae@172.30.0.191 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6397ms with no response
[2011-11-03 06:46:05] Retransmitting #7 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7a7c8ee1
Max-Forwards: 70
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
Contact: <sip:4118@172.30.0.10:5060>
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4118">
<address uri="sip:4118@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4127@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK89cc326e84D05EB9
From: "Office 105" <sip:4131@customer.com>;tag=4606BFFE-5F8266C9
To: <sip:4127@customer.com>;tag=as4b66acb0
CSeq: 4 SUBSCRIBE
Call-ID: 3000b8a7-86241b9a-f1b5fbc5@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4131", realm="asterisk", nonce="02430a8c", uri="sip:4127@172.30.0.10:5060", response="bdf22a8b7d2151ad5b45a0596c25cc26", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (16 headers 0 lines) ---
[2011-11-03 06:46:05] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:05]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK89cc326e84D05EB9;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=4606BFFE-5F8266C9
To: <sip:4127@customer.com>;tag=as4b66acb0
Call-ID: 3000b8a7-86241b9a-f1b5fbc5@172.30.0.170
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ca703e2
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as3e7e7001
To: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 61543496-3b870d21-3bdb7ca4@172.30.0.164 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6379ms with no response
[2011-11-03 06:46:05] Really destroying SIP dialog '61543496-3b870d21-3bdb7ca4@172.30.0.164' Method: SUBSCRIBE
[2011-11-03 06:46:05] Really destroying SIP dialog '3000b8a7-86241b9a-f1b5fbc5@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.166:5060 --->
SUBSCRIBE sip:4135@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKe418d91c985C71C5
From: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
To: <sip:4135@customer.com>;tag=as35e10d70
CSeq: 4 SUBSCRIBE
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4137", realm="asterisk", nonce="4c9e7660", uri="sip:4135@172.30.0.10:5060", response="486d452fdae44d799845f7e69cce3438", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (16 headers 0 lines) ---
[2011-11-03 06:46:05] Found peer '4137' for '4137' from 172.30.0.166:5060
[2011-11-03 06:46:05] Looking for 4135 in default-local (domain customer.com:5060)
[2011-11-03 06:46:05]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKe418d91c985C71C5;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
To: <sip:4135@customer.com>;tag=as35e10d70
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:05] set_destination: Parsing <sip:4137@172.30.0.166:5060> for address/port to send to
[2011-11-03 06:46:05] set_destination: set destination to 172.30.0.166:5060
[2011-11-03 06:46:05] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK579cb13c
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as35e10d70
To: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK695ee698
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36bb77a6
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as29d46f72
To: "4149" <sip:4149@customer.com>;tag=1A036AB2-15599CC1
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: c900a514-645b1ad3-24e6613a@172.30.0.188
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e443b0c
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as3e8cdda8
To: "4138" <sip:4138@customer.com>;tag=C87DED41-A6743BBA
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: e745cfbd-1ebc4c6-b065648b@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05] Retransmitting #5 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ffd7ca5
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as17a56026
To: "4122" <sip:4122@customer.com>;tag=411DC2D4-7DCDD2B
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 33aaa645-64e7904c-f13a5d23@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]
<--- SIP read from UDP:172.30.0.177:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK31dff7ddB397B5B2
From: "4135" <sip:4135@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4135@customer.com>
CSeq: 4 REGISTER
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4135", realm="asterisk", nonce="48003e5f", uri="sip:customer.com:5060", response="df9e0910edac681bed2c61aaea03c50c", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:05] --- (13 headers 0 lines) ---
[2011-11-03 06:46:05] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:46:05]     -- Unregistered SIP '4135'
[2011-11-03 06:46:05]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK31dff7ddB397B5B2;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4135@customer.com>;tag=as4e2f8472
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.177
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:05 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:05] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:46:05] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:46:05] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK189fae8d
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as661a5acc
To: "4118" <sip:4118@customer.com>;tag=8EB8376-7108A38B
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: cb40749-8afc3f9e-53672db3@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4118
[2011-11-03 06:46:05] set_destination: Parsing <sip:4140@172.30.0.175:5060> for address/port to send to
[2011-11-03 06:46:05] set_destination: set destination to 172.30.0.175:5060
[2011-11-03 06:46:05] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1a63e0fa
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as0a28bf84
To: "4140" <sip:4140@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 4ae23bb0-4380e1cb-431a995e@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4140
[2011-11-03 06:46:05] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:46:05] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:46:05] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f895944
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as01dea76e
To: "4124" <sip:4124@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: d7247dec-50473e67-484b175a@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4124
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:46:05] Scheduling destruction of SIP dialog 'b85bd9a5-c8e45da-e219d72b@172.30.0.177' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:46:05] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:46:05] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:46:05] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21f2ad9e
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4141
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6bf6199c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1acf8c29
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as58f1accb
To: "4168" <sip:4168@customer.com>;tag=93C331FA-B8037925
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK385dbd1c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as01d85198
To: "Leon Zhong" <sip:4134@customer.com>;tag=1C6DE198-7CE57F91
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 857aaf26-f301a987-4ad40894@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:46:05] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK33510b01
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as656896f7
To: "4148" <sip:4148@customer.com>;tag=13B5A662-4E562EC9
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: b1905049-17fabaa8-f17cb0bf@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:46:05]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:46:05] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ca703e2
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as3e7e7001
To: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:06] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:06] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:06] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25717256
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2024ac3b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as754e3632
To: "Office 131" <sip:4113@customer.com>;tag=B73F6B3E-E85CC275
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f02c36fd-a6d8f84-6155c2db@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:06]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:46:06] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:06] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:06] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a8a1e87
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4173
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK579cb13c
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as35e10d70
To: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Really destroying SIP dialog '9dd773e-6947e73b-b12100b4@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4120@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKc1ef79f8A365AEED
From: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
To: <sip:4120@customer.com>;tag=as203f8f34
CSeq: 3 SUBSCRIBE
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="4cd280e0", uri="sip:4120@customer.com:5060", response="751078c5328ce56da5a8b46af9f90f92", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (16 headers 0 lines) ---
[2011-11-03 06:46:06] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:46:06] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96'
[2011-11-03 06:46:06]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKc1ef79f8A365AEED;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
To: <sip:4120@customer.com>;tag=as203f8f34
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3179c43f", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:06] Scheduling destruction of SIP dialog '38a4be17-cb8f6e38-3efd882d@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK189fae8d
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as661a5acc
To: "4118" <sip:4118@customer.com>;tag=8EB8376-7108A38B
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: cb40749-8afc3f9e-53672db3@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1a63e0fa
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as0a28bf84
To: "4140" <sip:4140@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 4ae23bb0-4380e1cb-431a995e@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f895944
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as01dea76e
To: "4124" <sip:4124@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: d7247dec-50473e67-484b175a@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21f2ad9e
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45fd4e80
From: <sip:4115@customer.com>;tag=as5c6debc1
To: "4145" <sip:4145@customer.com>;tag=EA33C254-47D56619
CSeq: 105 NOTIFY
Call-ID: 50bbd9ab-e9c7e64-fc3994e9@172.30.0.167
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (10 headers 0 lines) ---
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a8a1e87
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK579cb13c
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as35e10d70
To: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14f4564b
From: <sip:4131@customer.com>;tag=as1343b7a2
To: "4136" <sip:4136@customer.com>;tag=F58CF41-E982E7DA
CSeq: 104 NOTIFY
Call-ID: d67fc766-e09267bf-51c6e6d8@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (11 headers 0 lines) ---
[2011-11-03 06:46:06] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK378ea673
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6e02a2c0
To: <sip:4140@172.30.0.175:5060>;tag=9E44804A-CAFB2FF5
CSeq: 102 OPTIONS
Call-ID: 039134d54c922e5279d167e65f157b75@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (9 headers 0 lines) ---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb9575a4448DC719B
From: "Office 131" <sip:4113@customer.com>;tag=BB7BAF6B-ECA88F92
To: <sip:4148@customer.com>;tag=as032151df
CSeq: 4 SUBSCRIBE
Call-ID: b9737532-4c0cc29-41dfcaf0@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="34aca8b9", uri="sip:4148@172.30.0.10:5060", response="22280b2fd7961f8c7049946a8ebd7701", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (16 headers 0 lines) ---
[2011-11-03 06:46:06] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:06]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb9575a4448DC719B;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=BB7BAF6B-ECA88F92
To: <sip:4148@customer.com>;tag=as032151df
Call-ID: b9737532-4c0cc29-41dfcaf0@172.30.0.195
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:06] Retransmitting #5 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK189fae8d
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as661a5acc
To: "4118" <sip:4118@customer.com>;tag=8EB8376-7108A38B
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: cb40749-8afc3f9e-53672db3@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Really destroying SIP dialog 'b9737532-4c0cc29-41dfcaf0@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4135@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa85a57f2BCA8E8EF
From: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
To: <sip:4135@customer.com>;tag=as48c3b1a3
CSeq: 4 SUBSCRIBE
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="1bb3528f", uri="sip:4135@172.30.0.10:5060", response="2b9609545fb501c68a195d2bf43f17af", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (16 headers 0 lines) ---
[2011-11-03 06:46:06] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:46:06] Looking for 4135 in default-local (domain customer.com:5060)
[2011-11-03 06:46:06]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa85a57f2BCA8E8EF;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
To: <sip:4135@customer.com>;tag=as48c3b1a3
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:06] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:46:06] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:46:06] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Really destroying SIP dialog '12ba535b-c1847776-8f1dad11@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1a63e0fa
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as0a28bf84
To: "4140" <sip:4140@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 4ae23bb0-4380e1cb-431a995e@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f895944
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as01dea76e
To: "4124" <sip:4124@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: d7247dec-50473e67-484b175a@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21f2ad9e
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #5 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK106817c6
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as58f94645
To: "4148" <sip:4148@customer.com>;tag=A421AA73-D0CB46A2
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8308066-263455ed-e9eab26c@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ca703e2
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as3e7e7001
To: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e45588d
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
CSeq: 104 NOTIFY
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (10 headers 0 lines) ---
[2011-11-03 06:46:06] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:06]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.181:5060
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a8a1e87
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.182:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c360bcd
From: <sip:4119@customer.com>;tag=as45b9714a
To: "Office 108" <sip:4128@customer.com>;tag=BFDCB33E-CD5B5BA7
CSeq: 103 NOTIFY
Call-ID: 1f0e8cd6-f0b74f7f-c83fed80@172.30.0.182
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (10 headers 0 lines) ---
[2011-11-03 06:46:06] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:06]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.182:5060
[2011-11-03 06:46:06] Really destroying SIP dialog '1f0e8cd6-f0b74f7f-c83fed80@172.30.0.182' Method: SUBSCRIBE
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04457a9c
From: <sip:4148@customer.com>;tag=as5cff5298
To: "4168" <sip:4168@customer.com>;tag=530ADCB5-91794658
CSeq: 104 NOTIFY
Call-ID: 59688f55-c01b9f8-f830e953@172.30.0.187
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (10 headers 0 lines) ---
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.170:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK402808bc
From: <sip:4127@customer.com>;tag=as4b66acb0
To: "Office 105" <sip:4131@customer.com>;tag=4606BFFE-5F8266C9
CSeq: 103 NOTIFY
Call-ID: 3000b8a7-86241b9a-f1b5fbc5@172.30.0.170
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (10 headers 0 lines) ---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4115@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKb8c42a18C00F1CD3
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
CSeq: 3 SUBSCRIBE
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="3a54b1f4", uri="sip:4115@customer.com:5060", response="bfc977b9d0a65817564f0c5eaac223f2", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (16 headers 0 lines) ---
[2011-11-03 06:46:06] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:06] Looking for 4115 in default-local (domain customer.com:5060)
[2011-11-03 06:46:06]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKb8c42a18C00F1CD3;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:06] set_destination: Parsing <sip:4171@172.30.0.165:5060> for address/port to send to
[2011-11-03 06:46:06] set_destination: set destination to 172.30.0.165:5060
[2011-11-03 06:46:06] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #5 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25717256
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #5 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0c35e750
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as54deba17
To: "4116" <sip:4116@customer.com>;tag=39B600A4-7C17D79D
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a99aa77a-c3145783-cd4773e4@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK579cb13c
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as35e10d70
To: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #7 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4130@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK318138e8CF75A3E3
From: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
To: <sip:4130@customer.com>;tag=as1443a841
CSeq: 3 SUBSCRIBE
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="5b7984bd", uri="sip:4130@customer.com:5060", response="9f89126a5c8d4bb71df39d7a9be8547e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (16 headers 0 lines) ---
[2011-11-03 06:46:06] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:46:06] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725'
[2011-11-03 06:46:06]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK318138e8CF75A3E3;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
To: <sip:4130@customer.com>;tag=as1443a841
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4321bc95", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:06] Scheduling destruction of SIP dialog 'dc8827b9-82fb4408-db62ad83@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK189fae8d
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as661a5acc
To: "4118" <sip:4118@customer.com>;tag=8EB8376-7108A38B
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: cb40749-8afc3f9e-53672db3@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1a63e0fa
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as0a28bf84
To: "4140" <sip:4140@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 4ae23bb0-4380e1cb-431a995e@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4171@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKb46015107CE91067
From: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
To: <sip:4171@customer.com>;tag=as2da31670
CSeq: 3 SUBSCRIBE
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="2ed92056", uri="sip:4171@customer.com:5060", response="5aa5164552232d39af1ae97d02401621", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (16 headers 0 lines) ---
[2011-11-03 06:46:06] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:46:06] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4173" <sip:4173@customer.com>;tag=497ECB75-910AE34'
[2011-11-03 06:46:06]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKb46015107CE91067;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
To: <sip:4171@customer.com>;tag=as2da31670
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b25517d", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:06] Scheduling destruction of SIP dialog '69de6be3-ca1cff92-22b56d79@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:06] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f895944
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as01dea76e
To: "4124" <sip:4124@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: d7247dec-50473e67-484b175a@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21f2ad9e
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a8a1e87
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.197:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1268fa28
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6ae6471f
To: <sip:4115@172.30.0.197:5060>;tag=F84E2B84-767901F
CSeq: 102 OPTIONS
Call-ID: 698c1e0c11b7a3e07b3a01166134be89@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (9 headers 0 lines) ---
[2011-11-03 06:46:06] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ecdcc2f
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as1fff0b57
To: "4136" <sip:4136@customer.com>;tag=F37A13B7-5AA7ACD0
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 39b92ead-60712846-68a8f09f@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.168:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK8a5390148883BB5D
From: "4125" <sip:4125@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4125@customer.com>
CSeq: 3 REGISTER
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Authorization: Digest username="4125", realm="asterisk", nonce="61e86105", uri="sip:customer.com:5060", response="f830840b8134d3c9ec1416bef0972f9e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (13 headers 0 lines) ---
[2011-11-03 06:46:06] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:46:06]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK8a5390148883BB5D;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4125@customer.com>;tag=as7cfa4124
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.168
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c7aecf3"
Content-Length: 0


<------------>
[2011-11-03 06:46:06] Scheduling destruction of SIP dialog 'b85bd9a5-c8e45da-e219d72b@172.30.0.168' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:06]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5d84ee
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0f964613
To: <sip:4168@172.30.0.187:5060>;tag=A114226A-27711F15
CSeq: 102 OPTIONS
Call-ID: 4bfef01f561aa11355bf50266675edc4@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:06] --- (9 headers 0 lines) ---
[2011-11-03 06:46:06] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK559240b6
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
CSeq: 106 NOTIFY
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (10 headers 0 lines) ---
[2011-11-03 06:46:07] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:07]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.181:5060
[2011-11-03 06:46:07] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45edadf2
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as616b4d7c
To: "4124" <sip:4124@customer.com>;tag=9798D95-C4ADE238
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4115@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKca688a1a9E4A6EFD
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
CSeq: 4 SUBSCRIBE
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="30b415df", uri="sip:4115@172.30.0.10:5060", response="5201f989dc8738c87c4cc430e54af462", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (16 headers 0 lines) ---
[2011-11-03 06:46:07] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:46:07] Looking for 4115 in default-local (domain customer.com:5060)
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ca703e2
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as3e7e7001
To: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4124@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK2fdbeb3eA9CAF585
From: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
To: <sip:4124@customer.com>;tag=as75491cdd
CSeq: 4 SUBSCRIBE
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="4c3e8f0a", uri="sip:4124@172.30.0.10:5060", response="8362eb2e3c0ea1ae81d16d7c33220394", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (16 headers 0 lines) ---
[2011-11-03 06:46:07] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:46:07] Looking for 4124 in default-local (domain customer.com:5060)
[2011-11-03 06:46:07]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK2fdbeb3eA9CAF585;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
To: <sip:4124@customer.com>;tag=as75491cdd
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:07] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:07] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:07] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK695ee698
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
CSeq: 106 NOTIFY
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (10 headers 0 lines) ---
[2011-11-03 06:46:07] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:07]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.178:5060
[2011-11-03 06:46:07] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #7 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4edcbecf
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a1a0149
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as5881d52c
To: "Office 106" <sip:4120@customer.com>;tag=D07B081B-73C9538A
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.180
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
CSeq: 104 NOTIFY
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (10 headers 0 lines) ---
[2011-11-03 06:46:07] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:07]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.175:5060
[2011-11-03 06:46:07] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK579cb13c
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as35e10d70
To: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Really destroying SIP dialog 'f78b7dd-72d575c0-4b416a5b@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4109@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKbcd7e9221C31E48D
From: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
To: <sip:4109@customer.com>;tag=as43fff8b0
CSeq: 3 SUBSCRIBE
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4131", realm="asterisk", nonce="08b0a00a", uri="sip:4109@customer.com:5060", response="43bef6cb8ea4f880ca9cee4eef9ccafc", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (16 headers 0 lines) ---
[2011-11-03 06:46:07] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:46:07] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850'
[2011-11-03 06:46:07]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKbcd7e9221C31E48D;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
To: <sip:4109@customer.com>;tag=as43fff8b0
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09b62cb5", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:07] Scheduling destruction of SIP dialog 'c6598b22-5d64568d-28d073f0@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK189fae8d
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as661a5acc
To: "4118" <sip:4118@customer.com>;tag=8EB8376-7108A38B
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: cb40749-8afc3f9e-53672db3@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e29496b
From: <sip:4141@customer.com>;tag=as0ebc6145
To: "4142" <sip:4142@customer.com>;tag=74D9A65D-DC6455DC
CSeq: 105 NOTIFY
Call-ID: f2d427b3-d153d31a-56517205@172.30.0.198
Contact: <sip:4142@172.30.0.198:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (11 headers 0 lines) ---
[2011-11-03 06:46:07] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:07] Retransmitting #5 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6bf6199c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1a63e0fa
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as0a28bf84
To: "4140" <sip:4140@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 4ae23bb0-4380e1cb-431a995e@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #5 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1acf8c29
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as58f1accb
To: "4168" <sip:4168@customer.com>;tag=93C331FA-B8037925
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f895944
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as01dea76e
To: "4124" <sip:4124@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: d7247dec-50473e67-484b175a@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK385dbd1c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as01d85198
To: "Leon Zhong" <sip:4134@customer.com>;tag=1C6DE198-7CE57F91
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 857aaf26-f301a987-4ad40894@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21f2ad9e
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK33510b01
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as656896f7
To: "4148" <sip:4148@customer.com>;tag=13B5A662-4E562EC9
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: b1905049-17fabaa8-f17cb0bf@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.188:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
CSeq: 107 NOTIFY
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (10 headers 0 lines) ---
[2011-11-03 06:46:07] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:07]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.188:5060
[2011-11-03 06:46:07] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25717256
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a8a1e87
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4135@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK427a3bf8EE1CB8D1
From: "Office 108" <sip:4128@customer.com>;tag=2B68B9DA-3214F2E3
To: <sip:4135@customer.com>;tag=as4e5800f1
CSeq: 4 SUBSCRIBE
Call-ID: 46c4bc42-3be57b0b-c015a82c@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4128", realm="asterisk", nonce="06169684", uri="sip:4135@172.30.0.10:5060", response="845c01c75e70ddca8a79645841d14462", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (16 headers 0 lines) ---
[2011-11-03 06:46:07] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:07]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK427a3bf8EE1CB8D1;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=2B68B9DA-3214F2E3
To: <sip:4135@customer.com>;tag=as4e5800f1
Call-ID: 46c4bc42-3be57b0b-c015a82c@172.30.0.182
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:07] Really destroying SIP dialog '46c4bc42-3be57b0b-c015a82c@172.30.0.182' Method: SUBSCRIBE
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4137@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKb3dbae6980E9507C
From: "4106" <sip:4106@customer.com>;tag=A545476A-F6881799
To: <sip:4137@customer.com>;tag=as0d102410
CSeq: 4 SUBSCRIBE
Call-ID: d365b45f-a8f7895a-46b66fc9@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4106", realm="asterisk", nonce="4c87ab1d", uri="sip:4137@172.30.0.10:5060", response="b0d7129012744c9e2359cfe66d85652e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (16 headers 0 lines) ---
[2011-11-03 06:46:07] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:07]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKb3dbae6980E9507C;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A545476A-F6881799
To: <sip:4137@customer.com>;tag=as0d102410
Call-ID: d365b45f-a8f7895a-46b66fc9@172.30.0.171
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:07] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b008ef1
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as08b6f44c
To: "4124" <sip:4124@customer.com>;tag=DAA75691-6EF42394
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: b5bf6aea-f1839b95-b22ea038@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #6 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3472001b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as2fb231f0
To: "4148" <sip:4148@customer.com>;tag=EE625669-435859C8
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: af04dc4c-fc6ea483-38848532@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a8e32c8
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as7ac82345
To: "Office 106" <sip:4120@customer.com>;tag=3F364E51-5B222F0
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 9f66077-d16d29c6-d057424d@172.30.0.180
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #6 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54c8630b
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as4f80e97d
To: "4136" <sip:4136@customer.com>;tag=EBBF8BD9-54CCB672
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: bef4b4f0-29e154c9-65f17362@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2a10e6db
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as15700ff6
To: "Leon Zhong" <sip:4134@customer.com>;tag=32A0D36-967AF257
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 7cdaf61-6dca67e6-6d8a2347@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Really destroying SIP dialog 'd365b45f-a8f7895a-46b66fc9@172.30.0.171' Method: SUBSCRIBE
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4115@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKca688a1a9E4A6EFD
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
CSeq: 4 SUBSCRIBE
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="30b415df", uri="sip:4115@172.30.0.10:5060", response="5201f989dc8738c87c4cc430e54af462", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (16 headers 0 lines) ---
[2011-11-03 06:46:07] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:07] Looking for 4115 in default-local (domain customer.com:5060)
[2011-11-03 06:46:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b5bf6aea-f1839b95-b22ea038@172.30.0.164 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6450ms with no response
[2011-11-03 06:46:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission f6adad1b-dff2c442-dcfbcfd1@172.30.0.188 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6446ms with no response
[2011-11-03 06:46:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission af04dc4c-fc6ea483-38848532@172.30.0.192 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6442ms with no response
[2011-11-03 06:46:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9f66077-d16d29c6-d057424d@172.30.0.180 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6439ms with no response
[2011-11-03 06:46:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission bef4b4f0-29e154c9-65f17362@172.30.0.172 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6436ms with no response
[2011-11-03 06:46:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 7cdaf61-6dca67e6-6d8a2347@172.30.0.179 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6433ms with no response
[2011-11-03 06:46:07] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e9e38c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b47e7b7
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fcdb7dc126fbbcd6189c39b358330de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.188:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41ee9e58
From: <sip:4122@customer.com>;tag=as313f76a4
To: "4149" <sip:4149@customer.com>;tag=3ED5E03B-8A44AA62
CSeq: 107 NOTIFY
Call-ID: 1182ed21-740a7c58-db9dd577@172.30.0.188
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (10 headers 0 lines) ---
[2011-11-03 06:46:07] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK266230cf
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as7c6d37d6
To: "4147" <sip:4147@customer.com>;tag=4C1D584F-2F9D61D0
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 3af79169-77fdc85a-8cf41d63@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK077564f8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f6f12ce
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 460c26e35e974dac4c5bc63c43be7ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.183:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62bd35b3
From: <sip:4138@customer.com>;tag=as5ed1c0df
To: "4147" <sip:4147@customer.com>;tag=CB4DDEC5-AA08E856
CSeq: 106 NOTIFY
Call-ID: c318f314-94094c8d-674f3cde@172.30.0.183
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (10 headers 0 lines) ---
[2011-11-03 06:46:07]
<--- SIP read from UDP:172.30.0.180:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a1a0149
From: <sip:4140@customer.com>;tag=as5881d52c
To: "Office 106" <sip:4120@customer.com>;tag=D07B081B-73C9538A
CSeq: 106 NOTIFY
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.180
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:07] --- (10 headers 0 lines) ---
[2011-11-03 06:46:07] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:07]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.180:5060
[2011-11-03 06:46:07] set_destination: Parsing <sip:4140@172.30.0.175:5060> for address/port to send to
[2011-11-03 06:46:07] set_destination: set destination to 172.30.0.175:5060
[2011-11-03 06:46:07] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:07] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.184:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
CSeq: 107 NOTIFY
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (10 headers 0 lines) ---
[2011-11-03 06:46:08] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:08]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.184:5060
[2011-11-03 06:46:08] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5225fcbd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50f1da3a
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0a78315778689242478ebb1150eab5c7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.183:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK266230cf
From: <sip:4106@customer.com>;tag=as7c6d37d6
To: "4147" <sip:4147@customer.com>;tag=4C1D584F-2F9D61D0
CSeq: 104 NOTIFY
Call-ID: 3af79169-77fdc85a-8cf41d63@172.30.0.183
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (10 headers 0 lines) ---
[2011-11-03 06:46:08] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:08]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.183:5060
[2011-11-03 06:46:08] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
CSeq: 104 NOTIFY
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (10 headers 0 lines) ---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4173@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKa9c56ada5E360275
From: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
To: <sip:4173@customer.com>;tag=as4cc49a90
CSeq: 3 SUBSCRIBE
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="710dcd8a", uri="sip:4173@customer.com:5060", response="b16423c2a72644668c4cb7eb57ac4d6f", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (16 headers 0 lines) ---
[2011-11-03 06:46:08] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:46:08] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04'
[2011-11-03 06:46:08]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKa9c56ada5E360275;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
To: <sip:4173@customer.com>;tag=as4cc49a90
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0cf3fbd5", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:08] Scheduling destruction of SIP dialog '191ae026-424cadc1-eaf44cdc@172.30.0.197' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK9ae561af225C79C
From: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
To: <sip:4148@customer.com>;tag=as5f466160
CSeq: 3 SUBSCRIBE
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="1430e5d6", uri="sip:4148@customer.com:5060", response="bff09c201297393403cb245435aa548f", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (16 headers 0 lines) ---
[2011-11-03 06:46:08] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:46:08] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6'
[2011-11-03 06:46:08]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK9ae561af225C79C;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
To: <sip:4148@customer.com>;tag=as5f466160
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="612585a5", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:08] Scheduling destruction of SIP dialog 'bc87819c-cb1e1045-a5c35ffa@172.30.0.177' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:08] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] Retransmitting #6 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK755a9481
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f882d2f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c079792
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1eb35a210a1486d8109bf10428ed556f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48ce56c6
From: <sip:4127@customer.com>;tag=as1a438c02
To: "4145" <sip:4145@customer.com>;tag=E42AA443-EC87CB1C
CSeq: 104 NOTIFY
Call-ID: bc992562-71188b8f-79764b78@172.30.0.167
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (10 headers 0 lines) ---
[2011-11-03 06:46:08] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff6cbdd
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as284d87e4
To: "4118" <sip:4118@customer.com>;tag=E9AFD17D-87EBC8D2
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 28129ff1-31250646-5c60a25b@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK04348c92
From: <sip:4141@customer.com>;tag=as6a30eab9
To: "4118" <sip:4118@customer.com>;tag=BF39822B-83BD6000
CSeq: 104 NOTIFY
Call-ID: a641b9eb-8a57e7c0-168deb55@172.30.0.173
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (10 headers 0 lines) ---
[2011-11-03 06:46:08] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:08]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.173:5060
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4122@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK6ac9ee192733A26E
From: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
To: <sip:4122@customer.com>;tag=as7ddb3c83
CSeq: 3 SUBSCRIBE
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="25e5cfd6", uri="sip:4122@customer.com:5060", response="e678860a07cf7a3ec623a4b10d722c12", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (16 headers 0 lines) ---
[2011-11-03 06:46:08] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:46:08] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493'
[2011-11-03 06:46:08]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK6ac9ee192733A26E;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
To: <sip:4122@customer.com>;tag=as7ddb3c83
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52cd05e5", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:08] Scheduling destruction of SIP dialog 'a8664db1-74778406-262df01b@172.30.0.173' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4109@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bKf5d4dc86B23F254B
From: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
To: <sip:4109@customer.com>;tag=as22382717
CSeq: 3 SUBSCRIBE
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="4d2143d0", uri="sip:4109@customer.com:5060", response="47276e463e13d3451b243ca5cb1f0d0d", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (16 headers 0 lines) ---
[2011-11-03 06:46:08] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:46:08] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829'
[2011-11-03 06:46:08]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bKf5d4dc86B23F254B;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
To: <sip:4109@customer.com>;tag=as22382717
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6bf7dcdc", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:08] Scheduling destruction of SIP dialog '623b63cb-a9e846dc-18385b89@172.30.0.163' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:08] Reliably Transmitting (no NAT) to 172.30.0.198:5060:
OPTIONS sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK118d53f7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6913496f
To: <sip:4142@172.30.0.198:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
CSeq: 105 NOTIFY
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (10 headers 0 lines) ---
[2011-11-03 06:46:08] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:08]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.167:5060
[2011-11-03 06:46:08] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] Retransmitting #5 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ca703e2
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as3e7e7001
To: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12349977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0414c506
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78d5c628706486467a0496f91be21c2c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:08]
<--- SIP read from UDP:172.30.0.195:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKbe1e56acE95C4583
From: "Office 131" <sip:4113@customer.com>;tag=6632B9A1-EBB6AE8
To: <sip:4113@customer.com>
CSeq: 4 REGISTER
Call-ID: b039439d-79868a24-bf6a4b7b@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4113", realm="asterisk", nonce="0c3bc11c", uri="sip:customer.com:5060", response="d36120b291f8662a4e7d116a468a52e3", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:08] --- (13 headers 0 lines) ---
[2011-11-03 06:46:08] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:46:08]     -- Unregistered SIP '4113'
[2011-11-03 06:46:08]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKbe1e56acE95C4583;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=6632B9A1-EBB6AE8
To: <sip:4113@customer.com>;tag=as780a9465
Call-ID: b039439d-79868a24-bf6a4b7b@172.30.0.195
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:08 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:08] Scheduling destruction of SIP dialog 'b039439d-79868a24-bf6a4b7b@172.30.0.195' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:08] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e9e38c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b47e7b7
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fcdb7dc126fbbcd6189c39b358330de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:08] Retransmitting #7 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK11ba61a3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1c439bdf
To: "Office 108" <sip:4128@customer.com>;tag=8B0D8457-86272A98
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 9355ea5-22367536-a032d0df@172.30.0.182
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 2ec5aa92-d2233079-7b3c7858@172.30.0.184 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6441ms with no response
[2011-11-03 06:46:08] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK077564f8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f6f12ce
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 460c26e35e974dac4c5bc63c43be7ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:08] Retransmitting #8 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK349b135e
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as29c5093b
To: "4136" <sip:4136@customer.com>;tag=AF9D5AC3-4C80CDC
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: 697be950-161e5129-5d1a37c2@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:08] Retransmitting #5 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK579cb13c
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as35e10d70
To: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.197:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK59dd4da6BF47DB41
From: "4115" <sip:4115@customer.com>;tag=FFA93B0D-23BE1028
To: <sip:4115@customer.com>
CSeq: 4 REGISTER
Call-ID: 5a7cf879-53c09394-e0e2c02f@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4115", realm="asterisk", nonce="7bcb5b6b", uri="sip:customer.com:5060", response="d9d34d1d8af6897d79e31a3471bb0338", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (13 headers 0 lines) ---
[2011-11-03 06:46:09] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:46:09]     -- Unregistered SIP '4115'
[2011-11-03 06:46:09]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK59dd4da6BF47DB41;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=FFA93B0D-23BE1028
To: <sip:4115@customer.com>;tag=as47295813
Call-ID: 5a7cf879-53c09394-e0e2c02f@172.30.0.197
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:09 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:09] Scheduling destruction of SIP dialog '5a7cf879-53c09394-e0e2c02f@172.30.0.197' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:09] Retransmitting #6 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36bb77a6
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as29d46f72
To: "4149" <sip:4149@customer.com>;tag=1A036AB2-15599CC1
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: c900a514-645b1ad3-24e6613a@172.30.0.188
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #6 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e443b0c
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as3e8cdda8
To: "4138" <sip:4138@customer.com>;tag=C87DED41-A6743BBA
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: e745cfbd-1ebc4c6-b065648b@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #6 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ffd7ca5
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as17a56026
To: "4122" <sip:4122@customer.com>;tag=411DC2D4-7DCDD2B
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 33aaa645-64e7904c-f13a5d23@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5225fcbd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50f1da3a
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0a78315778689242478ebb1150eab5c7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09] Retransmitting #7 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK183c55e2
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK189fae8d
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as661a5acc
To: "4118" <sip:4118@customer.com>;tag=8EB8376-7108A38B
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: cb40749-8afc3f9e-53672db3@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1a63e0fa
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as0a28bf84
To: "4140" <sip:4140@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 4ae23bb0-4380e1cb-431a995e@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f895944
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as01dea76e
To: "4124" <sip:4124@customer.com>;tag=286A7866-AE4EB571
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: d7247dec-50473e67-484b175a@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKd9883a5cCC392AF7
From: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
To: <sip:4148@customer.com>;tag=as314b084b
CSeq: 4 SUBSCRIBE
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="79135535", uri="sip:4148@172.30.0.10:5060", response="3bb4f76c542bc9517f5350ce7bacd171", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (16 headers 0 lines) ---
[2011-11-03 06:46:09] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:46:09] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:46:09]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKd9883a5cCC392AF7;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
To: <sip:4148@customer.com>;tag=as314b084b
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:09] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:46:09] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ee803e8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as314b084b
To: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c900a514-645b1ad3-24e6613a@172.30.0.188 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6434ms with no response
[2011-11-03 06:46:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e745cfbd-1ebc4c6-b065648b@172.30.0.163 for seqno 107 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6429ms with no response
[2011-11-03 06:46:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 33aaa645-64e7904c-f13a5d23@172.30.0.176 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6424ms with no response
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21f2ad9e
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a8a1e87
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #6 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2024ac3b
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as754e3632
To: "Office 131" <sip:4113@customer.com>;tag=B73F6B3E-E85CC275
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f02c36fd-a6d8f84-6155c2db@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
CSeq: 104 NOTIFY
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (10 headers 0 lines) ---
[2011-11-03 06:46:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission f02c36fd-a6d8f84-6155c2db@172.30.0.195 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6389ms with no response
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4127@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKe0d82023960F200
From: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
To: <sip:4127@customer.com>;tag=as1df1c6ea
CSeq: 6 SUBSCRIBE
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="6622e16f", uri="sip:4127@172.30.0.10:5060", response="dbc923df0cd8daf7ec789dc420026045", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (16 headers 0 lines) ---
[2011-11-03 06:46:09] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:46:09] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:46:09]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKe0d82023960F200;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
To: <sip:4127@customer.com>;tag=as1df1c6ea
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 6 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:09] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:46:09] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0414ad84
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 110 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b22cf05
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0d392b7e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 52e6b2c5667469ae70d82ced56d68523@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ee803e8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as314b084b
To: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKe9fd0b971E28A26C
From: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
To: <sip:4148@customer.com>;tag=as59b14bbd
CSeq: 4 SUBSCRIBE
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="217e30fe", uri="sip:4148@172.30.0.10:5060", response="f8fdc1f3406202ce4ada153394a3a384", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (16 headers 0 lines) ---
[2011-11-03 06:46:09] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:46:09] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:46:09]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKe9fd0b971E28A26C;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
To: <sip:4148@customer.com>;tag=as59b14bbd
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:09] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:46:09] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4136@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK9547b82aDAA39DDF
From: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
To: <sip:4136@customer.com>;tag=as1938f6a1
CSeq: 4 SUBSCRIBE
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="2ba40ead", uri="sip:4136@172.30.0.10:5060", response="9fb33584e186023030509ee910ac8776", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (16 headers 0 lines) ---
[2011-11-03 06:46:09] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:46:09] Looking for 4136 in default-local (domain customer.com:5060)
[2011-11-03 06:46:09]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK9547b82aDAA39DDF;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
To: <sip:4136@customer.com>;tag=as1938f6a1
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:09] set_destination: Parsing <sip:4138@172.30.0.163:5060> for address/port to send to
[2011-11-03 06:46:09] set_destination: set destination to 172.30.0.163:5060
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff03f7e
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1938f6a1
To: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0414ad84
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 110 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Really destroying SIP dialog 'a4ed9564-8af89fe9-41db734a@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK183c55e2
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
CSeq: 105 NOTIFY
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (10 headers 0 lines) ---
[2011-11-03 06:46:09] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:09]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.178:5060
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b6bb20b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1beafe92
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1751b05d567f0c7652d39df040463da9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ee803e8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as314b084b
To: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a8308066-263455ed-e9eab26c@172.30.0.192 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6427ms with no response
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f882d2f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c079792
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1eb35a210a1486d8109bf10428ed556f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25717256
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff03f7e
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1938f6a1
To: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0975328c
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as728e16e0
To: "4136" <sip:4136@customer.com>;tag=19F02E68-92FA0841
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: cc56bb5f-1e5b7278-c0f81851@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #7 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5c031c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0414ad84
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 110 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
OPTIONS sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: <sip:4136@172.30.0.172:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKe9fd0b971E28A26C
From: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
To: <sip:4148@customer.com>;tag=as59b14bbd
CSeq: 4 SUBSCRIBE
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="217e30fe", uri="sip:4148@172.30.0.10:5060", response="f8fdc1f3406202ce4ada153394a3a384", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (16 headers 0 lines) ---
[2011-11-03 06:46:09] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:09] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:46:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9355ea5-22367536-a032d0df@172.30.0.182 for seqno 107 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 16923ms with no response
[2011-11-03 06:46:09] Retransmitting #5 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #8 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7a7c8ee1
Max-Forwards: 70
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
Contact: <sip:4118@172.30.0.10:5060>
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4118">
<address uri="sip:4118@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.198:5060:
OPTIONS sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK118d53f7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6913496f
To: <sip:4142@172.30.0.198:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09] Retransmitting #6 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0c35e750
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as54deba17
To: "4116" <sip:4116@customer.com>;tag=39B600A4-7C17D79D
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a99aa77a-c3145783-cd4773e4@172.30.0.199
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff03f7e
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1938f6a1
To: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12349977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0414c506
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78d5c628706486467a0496f91be21c2c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4173@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK6f2551584EB88FF3
From: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
To: <sip:4173@customer.com>;tag=as63503940
CSeq: 4 SUBSCRIBE
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4108", realm="asterisk", nonce="002ff46a", uri="sip:4173@172.30.0.10:5060", response="e51763711eeba1d413a46b99a6f77d85", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:09] --- (16 headers 0 lines) ---
[2011-11-03 06:46:09] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:46:09] Looking for 4173 in default-local (domain customer.com:5060)
[2011-11-03 06:46:09]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK6f2551584EB88FF3;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
To: <sip:4173@customer.com>;tag=as63503940
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:09] set_destination: Parsing <sip:4108@172.30.0.196:5060> for address/port to send to
[2011-11-03 06:46:09] set_destination: set destination to 172.30.0.196:5060
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK309e3095
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as63503940
To: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e9e38c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b47e7b7
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fcdb7dc126fbbcd6189c39b358330de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a99aa77a-c3145783-cd4773e4@172.30.0.199 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6467ms with no response
[2011-11-03 06:46:09] Really destroying SIP dialog 'ed08d2df-9f1b6092-9f09697d@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:46:09] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ee803e8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as314b084b
To: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:09] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK077564f8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f6f12ce
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 460c26e35e974dac4c5bc63c43be7ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:09] set_destination: Parsing <sip:4128@172.30.0.182:5060> for address/port to send to
[2011-11-03 06:46:09] set_destination: set destination to 172.30.0.182:5060
[2011-11-03 06:46:09] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK70fde51f
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as1105be02
To: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4145@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK929c40e482CFFDDD
From: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
To: <sip:4145@customer.com>;tag=as5f5dbb11
CSeq: 4 SUBSCRIBE
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="7b40474c", uri="sip:4145@172.30.0.10:5060", response="3d73b039f80f6ca11ba62220c892f148", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:10] --- (16 headers 0 lines) ---
[2011-11-03 06:46:10] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:46:10] Looking for 4145 in default-local (domain customer.com:5060)
[2011-11-03 06:46:10]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK929c40e482CFFDDD;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
To: <sip:4145@customer.com>;tag=as5f5dbb11
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:10] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:46:10] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:46:10] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e66be5d
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as5f5dbb11
To: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK309e3095
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as63503940
To: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0414ad84
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 110 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Really destroying SIP dialog 'c318f314-94094c8d-674f3cde@172.30.0.183' Method: SUBSCRIBE
[2011-11-03 06:46:10] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK70fde51f
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as1105be02
To: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6329ms with no response
[2011-11-03 06:46:10] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5225fcbd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50f1da3a
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0a78315778689242478ebb1150eab5c7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Really destroying SIP dialog '8b9bc84d-4ae23bb0-4380e1cb@172.30.0.164' Method: SUBSCRIBE
[2011-11-03 06:46:10]
<--- SIP read from UDP:172.30.0.183:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bKb850279b2AF93A3C
From: "4147" <sip:4147@customer.com>;tag=3F243421-3F10D052
To: <sip:4147@customer.com>
CSeq: 3 REGISTER
Call-ID: c36bdc75-77e4d686-91aca9af@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4147", realm="asterisk", nonce="384d3c49", uri="sip:customer.com:5060", response="73392cea87047a7129537703a9a755c6", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:10] --- (13 headers 0 lines) ---
[2011-11-03 06:46:10] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:46:10]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bKb850279b2AF93A3C;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=3F243421-3F10D052
To: <sip:4147@customer.com>;tag=as530e21c5
Call-ID: c36bdc75-77e4d686-91aca9af@172.30.0.183
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29154efd"
Content-Length: 0


<------------>
[2011-11-03 06:46:10] Scheduling destruction of SIP dialog 'c36bdc75-77e4d686-91aca9af@172.30.0.183' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:10] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff03f7e
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1938f6a1
To: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK309e3095
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as63503940
To: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e66be5d
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as5f5dbb11
To: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK4ed404fb2CA5289C
From: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
To: <sip:4148@customer.com>;tag=as4b9f0e07
CSeq: 3 SUBSCRIBE
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4147", realm="asterisk", nonce="71472f4d", uri="sip:4148@customer.com:5060", response="0239b267401556c294b32d19667b12ed", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:10] --- (16 headers 0 lines) ---
[2011-11-03 06:46:10] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:46:10] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A'
[2011-11-03 06:46:10]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK4ed404fb2CA5289C;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
To: <sip:4148@customer.com>;tag=as4b9f0e07
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="362eb861", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:10] Scheduling destruction of SIP dialog 'cf08829-f486811a-8885c823@172.30.0.183' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:10] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK70fde51f
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as1105be02
To: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4136@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK9547b82aDAA39DDF
From: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
To: <sip:4136@customer.com>;tag=as1938f6a1
CSeq: 4 SUBSCRIBE
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="2ba40ead", uri="sip:4136@172.30.0.10:5060", response="9fb33584e186023030509ee910ac8776", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:10] --- (16 headers 0 lines) ---
[2011-11-03 06:46:10] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:10] Looking for 4136 in default-local (domain customer.com:5060)
[2011-11-03 06:46:10] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b22cf05
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0d392b7e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 52e6b2c5667469ae70d82ced56d68523@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e66be5d
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as5f5dbb11
To: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Really destroying SIP dialog 'bc992562-71188b8f-79764b78@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:46:10] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f882d2f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c079792
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1eb35a210a1486d8109bf10428ed556f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Really destroying SIP dialog '7685f52c-15168611-bf30b852@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:46:10] Retransmitting #8 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK58e487ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as772fc164
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0dbe4e0145ccbdd427e1d5cd38b6b51d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK309e3095
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as63503940
To: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10]
<--- SIP read from UDP:172.30.0.192:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK3f2c7d2e4072A2F5
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>
CSeq: 3 REGISTER
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4148", realm="asterisk", nonce="26cd763e", uri="sip:customer.com:5060", response="da248491a7eee044cb097f05981ce239", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:10] --- (13 headers 0 lines) ---
[2011-11-03 06:46:10] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:46:10]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK3f2c7d2e4072A2F5;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>;tag=as431ec1e5
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21bc3c43"
Content-Length: 0


<------------>
[2011-11-03 06:46:10] Scheduling destruction of SIP dialog '9e5886d1-1a3bbf70-bd6245c7@172.30.0.192' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:10] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b6bb20b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1beafe92
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1751b05d567f0c7652d39df040463da9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
OPTIONS sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: <sip:4136@172.30.0.172:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK70fde51f
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as1105be02
To: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ee803e8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as314b084b
To: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9be0f5d8-a82d0833-1c4d0606@172.30.0.175 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6409ms with no response
[2011-11-03 06:46:10] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9be0f5d8-a82d0833-1c4d0606@172.30.0.187 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6402ms with no response
[2011-11-03 06:46:10] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK385dbd1c
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as01d85198
To: "Leon Zhong" <sip:4134@customer.com>;tag=1C6DE198-7CE57F91
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 857aaf26-f301a987-4ad40894@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK082b9e9d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ab560bf
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4b67777e041dd836393636ec6afa3e86@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Retransmitting #2 (no NAT) to 172.30.0.198:5060:
OPTIONS sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK118d53f7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6913496f
To: <sip:4142@172.30.0.198:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b1905049-17fabaa8-f17cb0bf@172.30.0.192 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2011-11-03 06:46:10] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK484a0061
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e35fe8a
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f4e77ad005058fb2cb1feba5a3e22ac@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK371c1d9a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as048c45eb
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 54fcd2432666720340cb6f795fdc2fc8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12349977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0414c506
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78d5c628706486467a0496f91be21c2c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10]
<--- SIP read from UDP:172.30.0.192:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5c031c
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
CSeq: 105 NOTIFY
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:10] --- (10 headers 0 lines) ---
[2011-11-03 06:46:10] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:10]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.192:5060
[2011-11-03 06:46:10] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 857aaf26-f301a987-4ad40894@172.30.0.179 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6476ms with no response
[2011-11-03 06:46:10] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ecdcc2f
Max-Forwards: 70
From: <sip:4106@customer.com>;tag=as1fff0b57
To: "4136" <sip:4136@customer.com>;tag=F37A13B7-5AA7ACD0
Contact: <sip:4106@172.30.0.10:5060>
Call-ID: 39b92ead-60712846-68a8f09f@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4106">
<address uri="sip:4106@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #4 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0414ad84
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 110 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e66be5d
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as5f5dbb11
To: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:10] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e9e38c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b47e7b7
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fcdb7dc126fbbcd6189c39b358330de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:10]
<--- SIP read from UDP:172.30.0.188:5060 --->
SUBSCRIBE sip:4123@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK2e70fb8cBC8628B
From: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
To: <sip:4123@customer.com>;tag=as4a9a0381
CSeq: 3 SUBSCRIBE
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
Contact: <sip:4149@172.30.0.188:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4149", realm="asterisk", nonce="0c216058", uri="sip:4123@customer.com:5060", response="6d3e85a3129e9b530c10fbf0d939a4d3", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:10] --- (16 headers 0 lines) ---
[2011-11-03 06:46:10] Found peer '4149' for '4149' from 172.30.0.188:5060
[2011-11-03 06:46:10] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE'
[2011-11-03 06:46:10]
<--- Transmitting (no NAT) to 172.30.0.188:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK2e70fb8cBC8628B;received=172.30.0.188
From: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
To: <sip:4123@customer.com>;tag=as4a9a0381
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0aff066a", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:10] Scheduling destruction of SIP dialog '3a4a0277-34a32d3e-ce93ab6d@172.30.0.188' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:11] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:11] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK077564f8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f6f12ce
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 460c26e35e974dac4c5bc63c43be7ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff03f7e
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1938f6a1
To: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:11] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5225fcbd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50f1da3a
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0a78315778689242478ebb1150eab5c7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Retransmitting #5 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4690f411
From: <sip:4108@customer.com>;tag=as23cbc3e8
To: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
CSeq: 105 NOTIFY
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (10 headers 0 lines) ---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.195:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKbe1e56acE95C4583
From: "Office 131" <sip:4113@customer.com>;tag=6632B9A1-EBB6AE8
To: <sip:4113@customer.com>
CSeq: 4 REGISTER
Call-ID: b039439d-79868a24-bf6a4b7b@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4113", realm="asterisk", nonce="0c3bc11c", uri="sip:customer.com:5060", response="d36120b291f8662a4e7d116a468a52e3", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (13 headers 0 lines) ---
[2011-11-03 06:46:11] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:46:11]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKbe1e56acE95C4583;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=6632B9A1-EBB6AE8
To: <sip:4113@customer.com>;tag=as780a9465
Call-ID: b039439d-79868a24-bf6a4b7b@172.30.0.195
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:11 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:11] Scheduling destruction of SIP dialog 'b039439d-79868a24-bf6a4b7b@172.30.0.195' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa82ca9931E6EE6C2
From: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
To: <sip:4148@customer.com>;tag=as5fc896fe
CSeq: 3 SUBSCRIBE
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="11aed63e", uri="sip:4148@customer.com:5060", response="e79325971b6e58038b41fc57db89fceb", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (16 headers 0 lines) ---
[2011-11-03 06:46:11] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:46:11] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082'
[2011-11-03 06:46:11]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa82ca9931E6EE6C2;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
To: <sip:4148@customer.com>;tag=as5fc896fe
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51ebb6eb", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:11] Scheduling destruction of SIP dialog '3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:11] Retransmitting #8 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4edcbecf
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:11] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b22cf05
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0d392b7e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 52e6b2c5667469ae70d82ced56d68523@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4120@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKf0b50fee75339D61
From: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
To: <sip:4120@customer.com>;tag=as6f074e76
CSeq: 3 SUBSCRIBE
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="2e6b239d", uri="sip:4120@customer.com:5060", response="de13ea333a3d9d4a6a7eec3854a51e01", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (16 headers 0 lines) ---
[2011-11-03 06:46:11] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:46:11] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005'
[2011-11-03 06:46:11]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKf0b50fee75339D61;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
To: <sip:4120@customer.com>;tag=as6f074e76
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e7166ca", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:11] Scheduling destruction of SIP dialog '57d0a5f6-6d5ab409-9e569698@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a8a1e87
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
CSeq: 104 NOTIFY
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (10 headers 0 lines) ---
[2011-11-03 06:46:11] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:11]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.178:5060
[2011-11-03 06:46:11] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK309e3095
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as63503940
To: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4116@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKd7c2e9de5F2AF125
From: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
To: <sip:4116@customer.com>;tag=as1b71ddfa
CSeq: 3 SUBSCRIBE
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="09d9297b", uri="sip:4116@customer.com:5060", response="22b7db4c8e3ed3d000f25fb296bf91dc", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (16 headers 0 lines) ---
[2011-11-03 06:46:11] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:46:11] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715'
[2011-11-03 06:46:11]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKd7c2e9de5F2AF125;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
To: <sip:4116@customer.com>;tag=as1b71ddfa
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b15b1cd", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:11] Scheduling destruction of SIP dialog '549cc804-ea7523fb-947f666a@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:11] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK70fde51f
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as1105be02
To: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:11] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f882d2f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c079792
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1eb35a210a1486d8109bf10428ed556f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.184:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKffacc2818D4514A0
From: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
To: <sip:4148@customer.com>;tag=as2c27a83d
CSeq: 3 SUBSCRIBE
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4123", realm="asterisk", nonce="42d8331e", uri="sip:4148@customer.com:5060", response="5870e043392117e93252cd74e638f073", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (16 headers 0 lines) ---
[2011-11-03 06:46:11] Found peer '4123' for '4123' from 172.30.0.184:5060
[2011-11-03 06:46:11] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4123" <sip:4123@customer.com>;tag=3E636E08-D924771F'
[2011-11-03 06:46:11]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKffacc2818D4514A0;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
To: <sip:4148@customer.com>;tag=as2c27a83d
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74b2ae53", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:11] Scheduling destruction of SIP dialog 'ea8b0c-8a3eb943-f9c5aff2@172.30.0.184' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e44e5e8
From: <sip:4140@customer.com>;tag=as73332664
To: "4140" <sip:4140@customer.com>;tag=7F408EDB-6540CCEE
CSeq: 104 NOTIFY
Call-ID: f78b7dd-72d575c0-4b416a5b@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (10 headers 0 lines) ---
[2011-11-03 06:46:11] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e66be5d
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as5f5dbb11
To: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:11] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b6bb20b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1beafe92
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1751b05d567f0c7652d39df040463da9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
OPTIONS sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: <sip:4136@172.30.0.172:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
CSeq: 105 NOTIFY
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (10 headers 0 lines) ---
[2011-11-03 06:46:11] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:11]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.181:5060
[2011-11-03 06:46:11] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK58e487ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as772fc164
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0dbe4e0145ccbdd427e1d5cd38b6b51d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Retransmitting #3 (no NAT) to 172.30.0.198:5060:
OPTIONS sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK118d53f7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6913496f
To: <sip:4142@172.30.0.198:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.164:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKb8da83d1F8A212D4
From: "4124" <sip:4124@customer.com>;tag=5F04DB7D-7F9DC260
To: <sip:4124@customer.com>
CSeq: 3 REGISTER
Call-ID: c2f6c859-ff2f4a9c-155d1a97@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
uthorization: Digest username="4124", realm="asterisk", nonce="4a6ef7b6", uri="sip:customer.com:5060", response="e78ffbffe094b81f506dfd927f9396b6", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (13 headers 0 lines) ---
[2011-11-03 06:46:11] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:46:11]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKb8da83d1F8A212D4;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=5F04DB7D-7F9DC260
To: <sip:4124@customer.com>;tag=as2c8eec9f
Call-ID: c2f6c859-ff2f4a9c-155d1a97@172.30.0.164
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1468ddde"
Content-Length: 0


<------------>
[2011-11-03 06:46:11] Scheduling destruction of SIP dialog 'c2f6c859-ff2f4a9c-155d1a97@172.30.0.164' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.168:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK8a5390148883BB5D
From: "4125" <sip:4125@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4125@customer.com>
CSeq: 3 REGISTER
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Authorization: Digest username="4125", realm="asterisk", nonce="61e86105", uri="sip:customer.com:5060", response="f830840b8134d3c9ec1416bef0972f9e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (13 headers 0 lines) ---
[2011-11-03 06:46:11] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:46:11]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK8a5390148883BB5D;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4125@customer.com>;tag=as7cfa4124
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.168
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:11 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:11] Scheduling destruction of SIP dialog 'b85bd9a5-c8e45da-e219d72b@172.30.0.168' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:11] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12349977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0414c506
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78d5c628706486467a0496f91be21c2c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK082b9e9d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ab560bf
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4b67777e041dd836393636ec6afa3e86@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
CSeq: 105 NOTIFY
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (10 headers 0 lines) ---
[2011-11-03 06:46:11] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK484a0061
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e35fe8a
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f4e77ad005058fb2cb1feba5a3e22ac@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e9e38c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b47e7b7
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fcdb7dc126fbbcd6189c39b358330de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Really destroying SIP dialog '2fcdb7dc126fbbcd6189c39b358330de@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:11] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK371c1d9a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as048c45eb
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 54fcd2432666720340cb6f795fdc2fc8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4136@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK9547b82aDAA39DDF
From: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
To: <sip:4136@customer.com>;tag=as1938f6a1
CSeq: 4 SUBSCRIBE
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="2ba40ead", uri="sip:4136@172.30.0.10:5060", response="9fb33584e186023030509ee910ac8776", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:11] --- (16 headers 0 lines) ---
[2011-11-03 06:46:11] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:11] Looking for 4136 in default-local (domain customer.com:5060)
[2011-11-03 06:46:11] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK077564f8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f6f12ce
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 460c26e35e974dac4c5bc63c43be7ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:11] Really destroying SIP dialog '460c26e35e974dac4c5bc63c43be7ff2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:11] Retransmitting #6 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ca703e2
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as3e7e7001
To: "4138" <sip:4138@customer.com>;tag=21B3DB71-C80AAB2A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 96cd4208-697b8485-5bf50d6e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.184:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08653a0b
From: <sip:4140@customer.com>;tag=as36949da8
To: "4123" <sip:4123@customer.com>;tag=F01CF0E5-7143EB24
CSeq: 107 NOTIFY
Call-ID: e7c3c399-a6f34c78-298c750f@172.30.0.184
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (10 headers 0 lines) ---
[2011-11-03 06:46:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 96cd4208-697b8485-5bf50d6e@172.30.0.163 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6407ms with no response
[2011-11-03 06:46:12] Really destroying SIP dialog '96cd4208-697b8485-5bf50d6e@172.30.0.163' Method: SUBSCRIBE
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4130@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK26e7409cCA7159A7
From: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
To: <sip:4130@customer.com>;tag=as1443a841
CSeq: 4 SUBSCRIBE
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="4321bc95", uri="sip:4130@172.30.0.10:5060", response="1f1c866d0c083ca000a2db6026e2286c", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (16 headers 0 lines) ---
[2011-11-03 06:46:12] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:46:12] Looking for 4130 in default-local (domain customer.com:5060)
[2011-11-03 06:46:12]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK26e7409cCA7159A7;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
To: <sip:4130@customer.com>;tag=as1443a841
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:12] set_destination: Parsing <sip:4171@172.30.0.165:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.165:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5225fcbd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50f1da3a
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0a78315778689242478ebb1150eab5c7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Really destroying SIP dialog '0a78315778689242478ebb1150eab5c7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4171@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKf95e9cc444E60EBB
From: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
To: <sip:4171@customer.com>;tag=as2da31670
CSeq: 4 SUBSCRIBE
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="3b25517d", uri="sip:4171@172.30.0.10:5060", response="7a8263e24dc3152a26b4a8d8238da260", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (16 headers 0 lines) ---
[2011-11-03 06:46:12] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:46:12] Looking for 4171 in default-local (domain customer.com:5060)
[2011-11-03 06:46:12]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKf95e9cc444E60EBB;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
To: <sip:4171@customer.com>;tag=as2da31670
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:12] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28a56265
Max-Forwards: 70
From: <sip:4171@customer.com>;tag=as2da31670
To: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
Contact: <sip:4171@172.30.0.10:5060>
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4171">
<address uri="sip:4171@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #6 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK579cb13c
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as35e10d70
To: "4137" <sip:4137@customer.com>;tag=9FE9D168-19556721
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c3800915-f35edc8a-d33941b@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ecdcc2f
From: <sip:4106@customer.com>;tag=as1fff0b57
To: "4136" <sip:4136@customer.com>;tag=F37A13B7-5AA7ACD0
CSeq: 104 NOTIFY
Call-ID: 39b92ead-60712846-68a8f09f@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (11 headers 0 lines) ---
[2011-11-03 06:46:12] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c3800915-f35edc8a-d33941b@172.30.0.166 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6378ms with no response
[2011-11-03 06:46:12] Really destroying SIP dialog 'c3800915-f35edc8a-d33941b@172.30.0.166' Method: SUBSCRIBE
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.168:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1e994644F1C601CD
From: "4125" <sip:4125@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4125@customer.com>
CSeq: 4 REGISTER
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Authorization: Digest username="4125", realm="asterisk", nonce="6c7aecf3", uri="sip:customer.com:5060", response="62f2a9e887469ea7ced32cf10bba73dc", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (13 headers 0 lines) ---
[2011-11-03 06:46:12] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:46:12]     -- Unregistered SIP '4125'
[2011-11-03 06:46:12]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1e994644F1C601CD;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=2ED306E9-DADBAACE
To: <sip:4125@customer.com>;tag=as7cfa4124
Call-ID: b85bd9a5-c8e45da-e219d72b@172.30.0.168
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:12 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:12] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57cddb12
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as78b92214
To: "4145" <sip:4145@customer.com>;tag=EE058870-AC96B105
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: c1501b8d-a832371e-7927619b@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4145
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:12] set_destination: Parsing <sip:4140@172.30.0.175:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.175:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6664aa31
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as50b7453c
To: "4140" <sip:4140@customer.com>;tag=671DAD64-A9BC9A9F
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4140
[2011-11-03 06:46:12] set_destination: Parsing <sip:4168@172.30.0.187:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.187:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086e517f
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as239c5cbb
To: "4168" <sip:4168@customer.com>;tag=65549956-704EFFE1
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 98d8ceb7-a724e02a-360d12d5@172.30.0.187
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Scheduling destruction of SIP dialog 'b85bd9a5-c8e45da-e219d72b@172.30.0.168' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4168
[2011-11-03 06:46:12] Retransmitting #5 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ee803e8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as314b084b
To: "4115" <sip:4115@customer.com>;tag=E90DCDE5-A12C2F00
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b06505b4-1b81c24f-6b7fd06a@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] set_destination: Parsing <sip:4122@172.30.0.176:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.176:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK39f89515
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as2de2594b
To: "4122" <sip:4122@customer.com>;tag=8E9DF161-47E876A8
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: b25beb44-5ab6b29b-925723c2@172.30.0.176
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4122
[2011-11-03 06:46:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission cb40749-8afc3f9e-53672db3@172.30.0.173 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6424ms with no response
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 4ae23bb0-4380e1cb-431a995e@172.30.0.175 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6419ms with no response
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:46:12] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1f0fab92
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:12] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68ccc3cb
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as76be7419
To: "Office 107" <sip:4141@customer.com>;tag=147C871-D3F5041A
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: de76fe84-3c752569-af99ea72@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4141
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28a56265
Max-Forwards: 70
From: <sip:4171@customer.com>;tag=as2da31670
To: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
Contact: <sip:4171@172.30.0.10:5060>
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4171">
<address uri="sip:4171@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission d7247dec-50473e67-484b175a@172.30.0.164 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6434ms with no response
[2011-11-03 06:46:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 945e8061-662c26ca-2cbdbea7@172.30.0.181 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6424ms with no response
[2011-11-03 06:46:12] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b22cf05
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0d392b7e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 52e6b2c5667469ae70d82ced56d68523@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Retransmitting #7 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK755a9481
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Really destroying SIP dialog '945e8061-662c26ca-2cbdbea7@172.30.0.181' Method: SUBSCRIBE
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
CSeq: 103 NOTIFY
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (10 headers 0 lines) ---
[2011-11-03 06:46:12] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:12]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.165:5060
[2011-11-03 06:46:12] Retransmitting #5 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0414ad84
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 110 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57cddb12
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as78b92214
To: "4145" <sip:4145@customer.com>;tag=EE058870-AC96B105
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: c1501b8d-a832371e-7927619b@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6664aa31
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as50b7453c
To: "4140" <sip:4140@customer.com>;tag=671DAD64-A9BC9A9F
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28a56265
Max-Forwards: 70
From: <sip:4171@customer.com>;tag=as2da31670
To: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
Contact: <sip:4171@172.30.0.10:5060>
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4171">
<address uri="sip:4171@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086e517f
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as239c5cbb
To: "4168" <sip:4168@customer.com>;tag=65549956-704EFFE1
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 98d8ceb7-a724e02a-360d12d5@172.30.0.187
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK39f89515
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as2de2594b
To: "4122" <sip:4122@customer.com>;tag=8E9DF161-47E876A8
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: b25beb44-5ab6b29b-925723c2@172.30.0.176
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0382b7fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5876808c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6fbd360316bd738864155aab6ccded69@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68ccc3cb
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as76be7419
To: "Office 107" <sip:4141@customer.com>;tag=147C871-D3F5041A
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: de76fe84-3c752569-af99ea72@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:46:12] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6731560f
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as203f8f34
To: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Really destroying SIP dialog '73da2ff0-edbeaf8b-16292a42@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fcc9962
From: <sip:4135@customer.com>;tag=as48c3b1a3
To: "Office 107" <sip:4141@customer.com>;tag=42C64B84-9A3CB669
CSeq: 105 NOTIFY
Call-ID: 945e8061-662c26ca-2cbdbea7@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (10 headers 0 lines) ---
[2011-11-03 06:46:12] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f882d2f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c079792
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1eb35a210a1486d8109bf10428ed556f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Really destroying SIP dialog '1eb35a210a1486d8109bf10428ed556f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:12] Retransmitting #5 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff03f7e
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1938f6a1
To: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff6cbdd
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as284d87e4
To: "4118" <sip:4118@customer.com>;tag=E9AFD17D-87EBC8D2
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 28129ff1-31250646-5c60a25b@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b6bb20b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1beafe92
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1751b05d567f0c7652d39df040463da9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.170:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK8a9f25d0BF0558EB
From: "Office 105" <sip:4131@customer.com>;tag=5F04DB7D-7F9DC260
To: <sip:4131@customer.com>
CSeq: 3 REGISTER
Call-ID: c2f6c859-ff2f4a9c-155d1a97@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4131", realm="asterisk", nonce="250b9f43", uri="sip:customer.com:5060", response="f12f0fc873520cc60e84bca9878ba99b", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (13 headers 0 lines) ---
[2011-11-03 06:46:12] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:46:12]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK8a9f25d0BF0558EB;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=5F04DB7D-7F9DC260
To: <sip:4131@customer.com>;tag=as35ee88b4
Call-ID: c2f6c859-ff2f4a9c-155d1a97@172.30.0.170
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d92923b"
Content-Length: 0


<------------>
[2011-11-03 06:46:12] Scheduling destruction of SIP dialog 'c2f6c859-ff2f4a9c-155d1a97@172.30.0.170' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:12] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6731560f
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as203f8f34
To: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
OPTIONS sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: <sip:4136@172.30.0.172:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c00c1dd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as400a9b8f
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1c08e29e519cfd604e2cfd126020ad30@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57cddb12
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as78b92214
To: "4145" <sip:4145@customer.com>;tag=EE058870-AC96B105
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: c1501b8d-a832371e-7927619b@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK58e487ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as772fc164
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0dbe4e0145ccbdd427e1d5cd38b6b51d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6664aa31
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as50b7453c
To: "4140" <sip:4140@customer.com>;tag=671DAD64-A9BC9A9F
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086e517f
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as239c5cbb
To: "4168" <sip:4168@customer.com>;tag=65549956-704EFFE1
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 98d8ceb7-a724e02a-360d12d5@172.30.0.187
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK39f89515
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as2de2594b
To: "4122" <sip:4122@customer.com>;tag=8E9DF161-47E876A8
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: b25beb44-5ab6b29b-925723c2@172.30.0.176
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Retransmitting #4 (no NAT) to 172.30.0.198:5060:
OPTIONS sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK118d53f7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6913496f
To: <sip:4142@172.30.0.198:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Really destroying SIP dialog '56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68ccc3cb
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as76be7419
To: "Office 107" <sip:4141@customer.com>;tag=147C871-D3F5041A
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: de76fe84-3c752569-af99ea72@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:12] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ccf5133
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10907ccf
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4ca07abb149a060e646073b860ccfeaa@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12349977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0414c506
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78d5c628706486467a0496f91be21c2c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Really destroying SIP dialog '78d5c628706486467a0496f91be21c2c@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:12]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
CSeq: 107 NOTIFY
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:12] --- (10 headers 0 lines) ---
[2011-11-03 06:46:12] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:12]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.178:5060
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK082b9e9d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ab560bf
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4b67777e041dd836393636ec6afa3e86@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:12] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK484a0061
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e35fe8a
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f4e77ad005058fb2cb1feba5a3e22ac@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK371c1d9a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as048c45eb
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 54fcd2432666720340cb6f795fdc2fc8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6731560f
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as203f8f34
To: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28a56265
Max-Forwards: 70
From: <sip:4171@customer.com>;tag=as2da31670
To: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
Contact: <sip:4171@172.30.0.10:5060>
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4171">
<address uri="sip:4171@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1f0fab92
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #9 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK349b135e
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as29c5093b
To: "4136" <sip:4136@customer.com>;tag=AF9D5AC3-4C80CDC
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: 697be950-161e5129-5d1a37c2@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #5 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK309e3095
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as63503940
To: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Really destroying SIP dialog 'bfbd9e39-ced80418-ec8d31af@172.30.0.178' Method: SUBSCRIBE
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
CSeq: 103 NOTIFY
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (10 headers 0 lines) ---
[2011-11-03 06:46:13] Really destroying SIP dialog '3eb75559-6dc219ac-50d94d13@172.30.0.171' Method: REGISTER
[2011-11-03 06:46:13] Retransmitting #5 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK70fde51f
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as1105be02
To: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57cddb12
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as78b92214
To: "4145" <sip:4145@customer.com>;tag=EE058870-AC96B105
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: c1501b8d-a832371e-7927619b@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6664aa31
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as50b7453c
To: "4140" <sip:4140@customer.com>;tag=671DAD64-A9BC9A9F
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086e517f
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as239c5cbb
To: "4168" <sip:4168@customer.com>;tag=65549956-704EFFE1
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 98d8ceb7-a724e02a-360d12d5@172.30.0.187
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK39f89515
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as2de2594b
To: "4122" <sip:4122@customer.com>;tag=8E9DF161-47E876A8
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: b25beb44-5ab6b29b-925723c2@172.30.0.176
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68ccc3cb
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as76be7419
To: "Office 107" <sip:4141@customer.com>;tag=147C871-D3F5041A
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: de76fe84-3c752569-af99ea72@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6623bd1d
From: <sip:4124@customer.com>;tag=as75491cdd
To: "4173" <sip:4173@customer.com>;tag=9A5DD2D9-9D265B8
CSeq: 107 NOTIFY
Call-ID: bfbd9e39-ced80418-ec8d31af@172.30.0.178
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (10 headers 0 lines) ---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4109@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK9436b75eC977AD29
From: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
To: <sip:4109@customer.com>;tag=as43fff8b0
CSeq: 4 SUBSCRIBE
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4131", realm="asterisk", nonce="09b62cb5", uri="sip:4109@172.30.0.10:5060", response="1469b5a9cf36c4a22164ca4996857fbc", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (16 headers 0 lines) ---
[2011-11-03 06:46:13] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:46:13] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:46:13]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK9436b75eC977AD29;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
To: <sip:4109@customer.com>;tag=as43fff8b0
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:13] set_destination: Parsing <sip:4131@172.30.0.170:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.170:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50859a83
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as43fff8b0
To: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #5 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e66be5d
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as5f5dbb11
To: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24ad2f4c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f104a05
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12ec8eed0280cad67a90293e3bd367be@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6731560f
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as203f8f34
To: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1acf8c29
From: <sip:4115@customer.com>;tag=as58f1accb
To: "4168" <sip:4168@customer.com>;tag=93C331FA-B8037925
CSeq: 104 NOTIFY
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.187
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (10 headers 0 lines) ---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b22cf05
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0d392b7e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 52e6b2c5667469ae70d82ced56d68523@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Really destroying SIP dialog '52e6b2c5667469ae70d82ced56d68523@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25717256
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
CSeq: 104 NOTIFY
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (11 headers 0 lines) ---
[2011-11-03 06:46:13] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50859a83
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as43fff8b0
To: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.164:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b008ef1
From: <sip:4113@customer.com>;tag=as08b6f44c
To: "4124" <sip:4124@customer.com>;tag=DAA75691-6EF42394
CSeq: 104 NOTIFY
Call-ID: b5bf6aea-f1839b95-b22ea038@172.30.0.164
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (10 headers 0 lines) ---
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21867acd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as017c026e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e91b4b704344d173a04045939e3b92a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.188:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK171f80b9
From: <sip:4113@customer.com>;tag=as7bcd1d12
To: "4149" <sip:4149@customer.com>;tag=82A9E24E-155C2DFD
CSeq: 104 NOTIFY
Call-ID: f6adad1b-dff2c442-dcfbcfd1@172.30.0.188
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (10 headers 0 lines) ---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.170:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0e9e38c6
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b47e7b7
To: <sip:4131@172.30.0.170:5060>;tag=1D0E7BEC-ABE96C67
CSeq: 102 OPTIONS
Call-ID: 2fcdb7dc126fbbcd6189c39b358330de@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (9 headers 0 lines) ---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0382b7fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5876808c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6fbd360316bd738864155aab6ccded69@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #6 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0975328c
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as728e16e0
To: "4136" <sip:4136@customer.com>;tag=19F02E68-92FA0841
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: cc56bb5f-1e5b7278-c0f81851@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50859a83
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as43fff8b0
To: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4134@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK5063c406DB4AD4E7
From: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
To: <sip:4134@customer.com>;tag=as40f08f92
CSeq: 3 SUBSCRIBE
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4134", realm="asterisk", nonce="61d24b12", uri="sip:4134@customer.com:5060", response="db5fdcc5ce87bb9bc463c4b737398a11", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (16 headers 0 lines) ---
[2011-11-03 06:46:13] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:46:13] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2'
[2011-11-03 06:46:13]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK5063c406DB4AD4E7;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
To: <sip:4134@customer.com>;tag=as40f08f92
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66508bce", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:13] Scheduling destruction of SIP dialog 'f385b0a3-50915880-1318c659@172.30.0.179' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4109@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK9436b75eC977AD29
From: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
To: <sip:4109@customer.com>;tag=as43fff8b0
CSeq: 4 SUBSCRIBE
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4131", realm="asterisk", nonce="09b62cb5", uri="sip:4109@172.30.0.10:5060", response="1469b5a9cf36c4a22164ca4996857fbc", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (16 headers 0 lines) ---
[2011-11-03 06:46:13] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:13] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b6bb20b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1beafe92
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1751b05d567f0c7652d39df040463da9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4128' is now UNREACHABLE!  Last qualify: 61
[2011-11-03 06:46:13] Really destroying SIP dialog '1751b05d567f0c7652d39df040463da9@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
OPTIONS sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: <sip:4136@172.30.0.172:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:46:13] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4136' is now UNREACHABLE!  Last qualify: 63
[2011-11-03 06:46:13] set_destination: Parsing <sip:4140@172.30.0.175:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.175:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16c493ae
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as345da678
To: "4140" <sip:4140@customer.com>;tag=8B547CAC-63405B27
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 4f2cbab3-e701fc86-daa75691@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4140
[2011-11-03 06:46:13] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c89e9ca
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as1124fe55
To: "4115" <sip:4115@customer.com>;tag=4E3134F3-82CD450E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: ad81a9a0-c02e6c3b-7b878056@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4115
[2011-11-03 06:46:13] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9f9f08
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as36b775b2
To: "Leon Zhong" <sip:4134@customer.com>;tag=D51BAD8B-9D0E2D48
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 31e9b6e0-cb9bdd39-346616de@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4134
[2011-11-03 06:46:13] set_destination: Parsing <sip:4123@172.30.0.184:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.184:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f40698
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as19cd3a67
To: "4123" <sip:4123@customer.com>;tag=A30FF0DF-8006EF6E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: a52898eb-eb76ceda-61e24101@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4123
[2011-11-03 06:46:13] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:13] Really destroying SIP dialog '6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4128[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:46:13] set_destination: Parsing <sip:4123@172.30.0.184:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.184:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4123
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4128 (queued)
[2011-11-03 06:46:13] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4173
[2011-11-03 06:46:13] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:46:13] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:46:13] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1855b0e9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1cd45702
To: "Leon Zhong" <sip:4134@customer.com>;tag=237BEF3D-2478B192
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 8722abbe-653b03f-3d5d56ec@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4134
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:46:13]   == Extension Changed 4136[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK58e487ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as772fc164
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0dbe4e0145ccbdd427e1d5cd38b6b51d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #9 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7a7c8ee1
Max-Forwards: 70
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
Contact: <sip:4118@172.30.0.10:5060>
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4118">
<address uri="sip:4118@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28a56265
Max-Forwards: 70
From: <sip:4171@customer.com>;tag=as2da31670
To: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
Contact: <sip:4171@172.30.0.10:5060>
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4171">
<address uri="sip:4171@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
CSeq: 106 NOTIFY
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (10 headers 0 lines) ---
[2011-11-03 06:46:13] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:13]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.175:5060
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK082b9e9d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ab560bf
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4b67777e041dd836393636ec6afa3e86@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c00c1dd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as400a9b8f
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1c08e29e519cfd604e2cfd126020ad30@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK484a0061
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e35fe8a
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f4e77ad005058fb2cb1feba5a3e22ac@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK371c1d9a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as048c45eb
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 54fcd2432666720340cb6f795fdc2fc8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ccf5133
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10907ccf
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4ca07abb149a060e646073b860ccfeaa@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16c493ae
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as345da678
To: "4140" <sip:4140@customer.com>;tag=8B547CAC-63405B27
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 4f2cbab3-e701fc86-daa75691@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c89e9ca
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as1124fe55
To: "4115" <sip:4115@customer.com>;tag=4E3134F3-82CD450E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: ad81a9a0-c02e6c3b-7b878056@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9f9f08
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as36b775b2
To: "Leon Zhong" <sip:4134@customer.com>;tag=D51BAD8B-9D0E2D48
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 31e9b6e0-cb9bdd39-346616de@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f40698
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as19cd3a67
To: "4123" <sip:4123@customer.com>;tag=A30FF0DF-8006EF6E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: a52898eb-eb76ceda-61e24101@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57cddb12
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as78b92214
To: "4145" <sip:4145@customer.com>;tag=EE058870-AC96B105
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: c1501b8d-a832371e-7927619b@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
CSeq: 106 NOTIFY
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:13] --- (10 headers 0 lines) ---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6664aa31
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as50b7453c
To: "4140" <sip:4140@customer.com>;tag=671DAD64-A9BC9A9F
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086e517f
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as239c5cbb
To: "4168" <sip:4168@customer.com>;tag=65549956-704EFFE1
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 98d8ceb7-a724e02a-360d12d5@172.30.0.187
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK39f89515
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as2de2594b
To: "4122" <sip:4122@customer.com>;tag=8E9DF161-47E876A8
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: b25beb44-5ab6b29b-925723c2@172.30.0.176
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1f0fab92
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1855b0e9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1cd45702
To: "Leon Zhong" <sip:4134@customer.com>;tag=237BEF3D-2478B192
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 8722abbe-653b03f-3d5d56ec@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:13] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68ccc3cb
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as76be7419
To: "Office 107" <sip:4141@customer.com>;tag=147C871-D3F5041A
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: de76fe84-3c752569-af99ea72@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK753d584e
From: <sip:4115@customer.com>;tag=as2574012d
To: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
CSeq: 103 NOTIFY
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (10 headers 0 lines) ---
[2011-11-03 06:46:14] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50859a83
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as43fff8b0
To: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4171@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKf95e9cc444E60EBB
From: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
To: <sip:4171@customer.com>;tag=as2da31670
CSeq: 4 SUBSCRIBE
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="3b25517d", uri="sip:4171@172.30.0.10:5060", response="7a8263e24dc3152a26b4a8d8238da260", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (16 headers 0 lines) ---
[2011-11-03 06:46:14] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:14] Looking for 4171 in default-local (domain customer.com:5060)
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16c493ae
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as345da678
To: "4140" <sip:4140@customer.com>;tag=8B547CAC-63405B27
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 4f2cbab3-e701fc86-daa75691@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6731560f
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as203f8f34
To: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c89e9ca
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as1124fe55
To: "4115" <sip:4115@customer.com>;tag=4E3134F3-82CD450E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: ad81a9a0-c02e6c3b-7b878056@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9f9f08
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as36b775b2
To: "Leon Zhong" <sip:4134@customer.com>;tag=D51BAD8B-9D0E2D48
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 31e9b6e0-cb9bdd39-346616de@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f40698
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as19cd3a67
To: "4123" <sip:4123@customer.com>;tag=A30FF0DF-8006EF6E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: a52898eb-eb76ceda-61e24101@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4127@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKe0d82023960F200
From: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
To: <sip:4127@customer.com>;tag=as1df1c6ea
CSeq: 6 SUBSCRIBE
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="6622e16f", uri="sip:4127@172.30.0.10:5060", response="dbc923df0cd8daf7ec789dc420026045", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (16 headers 0 lines) ---
[2011-11-03 06:46:14] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:14] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57acf09c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as035a47b9
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 105901fa1804429c08194de1570d6147@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1855b0e9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1cd45702
To: "Leon Zhong" <sip:4134@customer.com>;tag=237BEF3D-2478B192
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 8722abbe-653b03f-3d5d56ec@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK4300c1d921BBC6FE
From: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
To: <sip:4148@customer.com>;tag=as5f466160
CSeq: 4 SUBSCRIBE
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="612585a5", uri="sip:4148@172.30.0.10:5060", response="4b9ccd2d8aac78573e9200819c58fd95", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (16 headers 0 lines) ---
[2011-11-03 06:46:14] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:46:14] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:46:14]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK4300c1d921BBC6FE;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
To: <sip:4148@customer.com>;tag=as5f466160
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:14] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7563e7ee
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5f466160
To: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4136@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK8848c90874E6BBBD
From: "4145" <sip:4145@customer.com>;tag=196B7755-1C8E6946
To: <sip:4136@customer.com>;tag=as208ce64c
CSeq: 4 SUBSCRIBE
Call-ID: 9dd773e-6947e73b-b12100b4@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="585b8d5a", uri="sip:4136@172.30.0.10:5060", response="b24ab10af1fe2e96c245523030ed55bc", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (16 headers 0 lines) ---
[2011-11-03 06:46:14] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:14]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK8848c90874E6BBBD;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=196B7755-1C8E6946
To: <sip:4136@customer.com>;tag=as208ce64c
Call-ID: 9dd773e-6947e73b-b12100b4@172.30.0.167
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24ad2f4c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f104a05
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12ec8eed0280cad67a90293e3bd367be@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] Really destroying SIP dialog '9dd773e-6947e73b-b12100b4@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4109@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bKa6d0b446F288A60B
From: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
To: <sip:4109@customer.com>;tag=as22382717
CSeq: 4 SUBSCRIBE
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="6bf7dcdc", uri="sip:4109@172.30.0.10:5060", response="3db91115a9d142af3bf9689c2a2a10ec", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (16 headers 0 lines) ---
[2011-11-03 06:46:14] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:46:14] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:46:14]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bKa6d0b446F288A60B;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
To: <sip:4109@customer.com>;tag=as22382717
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:14] set_destination: Parsing <sip:4138@172.30.0.163:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.163:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c0e1037
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as22382717
To: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7563e7ee
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5f466160
To: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Really destroying SIP dialog '9be0f5d8-a82d0833-1c4d0606@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK118d53f7
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6913496f
To: "4142" <sip:4142@172.30.0.198:5060>;tag=345B318F-BC0E3206
CSeq: 102 OPTIONS
Call-ID: 56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060
Contact: <sip:4142@172.30.0.198:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (14 headers 0 lines) ---
[2011-11-03 06:46:14] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16c493ae
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as345da678
To: "4140" <sip:4140@customer.com>;tag=8B547CAC-63405B27
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 4f2cbab3-e701fc86-daa75691@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c89e9ca
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as1124fe55
To: "4115" <sip:4115@customer.com>;tag=4E3134F3-82CD450E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: ad81a9a0-c02e6c3b-7b878056@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9f9f08
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as36b775b2
To: "Leon Zhong" <sip:4134@customer.com>;tag=D51BAD8B-9D0E2D48
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 31e9b6e0-cb9bdd39-346616de@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f40698
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as19cd3a67
To: "4123" <sip:4123@customer.com>;tag=A30FF0DF-8006EF6E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: a52898eb-eb76ceda-61e24101@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e93dede
From: <sip:4115@customer.com>;tag=as4b34e14d
To: "4140" <sip:4140@customer.com>;tag=AA667848-38F99023
CSeq: 106 NOTIFY
Call-ID: 9be0f5d8-a82d0833-1c4d0606@172.30.0.175
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (10 headers 0 lines) ---
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21867acd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as017c026e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e91b4b704344d173a04045939e3b92a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c0e1037
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as22382717
To: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.180:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12349977
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0414c506
To: <sip:4120@172.30.0.180:5060>;tag=44E60EBB-56C7972A
CSeq: 102 OPTIONS
Call-ID: 78d5c628706486467a0496f91be21c2c@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (9 headers 0 lines) ---
[2011-11-03 06:46:14] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1855b0e9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1cd45702
To: "Leon Zhong" <sip:4134@customer.com>;tag=237BEF3D-2478B192
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 8722abbe-653b03f-3d5d56ec@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0382b7fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5876808c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6fbd360316bd738864155aab6ccded69@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7563e7ee
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5f466160
To: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #9 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.167:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK2fb656dcEF6C1201
From: "4145" <sip:4145@customer.com>;tag=7EDD76B0-8C141245
To: <sip:4145@customer.com>
CSeq: 3 REGISTER
Call-ID: 4750d9ac-8361b091-62b430d2@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4145", realm="asterisk", nonce="6ad6acd8", uri="sip:customer.com:5060", response="b62c39169c3be1dc34a65c4ac475408b", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (13 headers 0 lines) ---
[2011-11-03 06:46:14] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:46:14]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK2fb656dcEF6C1201;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=7EDD76B0-8C141245
To: <sip:4145@customer.com>;tag=as3b4a395b
Call-ID: 4750d9ac-8361b091-62b430d2@172.30.0.167
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0747e58d"
Content-Length: 0


<------------>
[2011-11-03 06:46:14] Scheduling destruction of SIP dialog '4750d9ac-8361b091-62b430d2@172.30.0.167' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.175:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK9436b75eC977AD29
From: "4140" <sip:4140@customer.com>;tag=5F04DB7D-7F9DC260
To: <sip:4140@customer.com>
CSeq: 3 REGISTER
Call-ID: c2f6c859-ff2f4a9c-155d1a97@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4140", realm="asterisk", nonce="0fe53cb2", uri="sip:customer.com:5060", response="4fa4c18ec5638661296df26f0e0e6cd2", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (13 headers 0 lines) ---
[2011-11-03 06:46:14] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:46:14]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK9436b75eC977AD29;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=5F04DB7D-7F9DC260
To: <sip:4140@customer.com>;tag=as7e82bef3
Call-ID: c2f6c859-ff2f4a9c-155d1a97@172.30.0.175
CSeq: 3 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a19224d"
Content-Length: 0


<------------>
[2011-11-03 06:46:14] Scheduling destruction of SIP dialog 'c2f6c859-ff2f4a9c-155d1a97@172.30.0.175' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:14] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK58e487ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as772fc164
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0dbe4e0145ccbdd427e1d5cd38b6b51d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e0d9b14
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as4cc49a90
To: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4123' is now UNREACHABLE!  Last qualify: 62
[2011-11-03 06:46:14] Really destroying SIP dialog '0dbe4e0145ccbdd427e1d5cd38b6b51d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c0e1037
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as22382717
To: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:46:14] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6321ddca
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as3de03838
To: "Office 107" <sip:4141@customer.com>;tag=8D7D9465-CD50C81E
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: af99d365-7ab2b31e-85d1028b@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4141
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4108 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:14] set_destination: Parsing <sip:4122@172.30.0.176:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.176:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f68fbe0
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as406b866e
To: "4122" <sip:4122@customer.com>;tag=20E6B604-928E615B
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 1692971-a61359b8-7bc438cf@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4122
[2011-11-03 06:46:14] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ed63e56
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as2d54b824
To: "4124" <sip:4124@customer.com>;tag=4834CBBA-59BBA8E5
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4124
[2011-11-03 06:46:14] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dfbc31d
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as08313b34
To: "4147" <sip:4147@customer.com>;tag=9A4A9A1D-A04807EE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3ebc8d41-ac62f472-8541dfbb@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4147
[2011-11-03 06:46:14] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:14] set_destination: Parsing <sip:4128@172.30.0.182:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.182:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fdcbd31
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as5b80a547
To: "Office 108" <sip:4128@customer.com>;tag=95EECC8A-82406E13
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 58f32bbe-4088e027-ca4039e8@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4123[default-local] new state Unavailable for Notify User 4128
[2011-11-03 06:46:14]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4141@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK1d750f46860373D3
From: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
To: <sip:4141@customer.com>;tag=as45e9cdb6
CSeq: 3 SUBSCRIBE
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="7b72d84a", uri="sip:4141@customer.com:5060", response="0262bf39ebf3e357936cf38fa9608b7f", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:14] --- (16 headers 0 lines) ---
[2011-11-03 06:46:14] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:46:14] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B'
[2011-11-03 06:46:14]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK1d750f46860373D3;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
To: <sip:4141@customer.com>;tag=as45e9cdb6
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3db62f34", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:14] Scheduling destruction of SIP dialog '12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50859a83
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as43fff8b0
To: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK082b9e9d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ab560bf
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4b67777e041dd836393636ec6afa3e86@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4108' is now UNREACHABLE!  Last qualify: 55
[2011-11-03 06:46:14] Really destroying SIP dialog '4b67777e041dd836393636ec6afa3e86@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4128
[2011-11-03 06:46:14] set_destination: Parsing <sip:4122@172.30.0.176:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.176:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07328dc3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as755a895e
To: "4122" <sip:4122@customer.com>;tag=962E0950-36450E7
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: ec30531d-f8af41a4-bdb82afb@172.30.0.176
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4122
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:14] set_destination: Parsing <sip:4149@172.30.0.188:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.188:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4149
[2011-11-03 06:46:14] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4148
[2011-11-03 06:46:14] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4173
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4108[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c00c1dd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as400a9b8f
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1c08e29e519cfd604e2cfd126020ad30@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK484a0061
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e35fe8a
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f4e77ad005058fb2cb1feba5a3e22ac@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] Really destroying SIP dialog '6f4e77ad005058fb2cb1feba5a3e22ac@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e0d9b14
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as4cc49a90
To: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK371c1d9a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as048c45eb
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 54fcd2432666720340cb6f795fdc2fc8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4148' is now UNREACHABLE!  Last qualify: 60
[2011-11-03 06:46:14] Really destroying SIP dialog '54fcd2432666720340cb6f795fdc2fc8@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4116 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:46:14] set_destination: Parsing <sip:4128@172.30.0.182:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.182:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4128
[2011-11-03 06:46:14] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2acf54bc
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4147
[2011-11-03 06:46:14] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4173
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4125 (queued)
[2011-11-03 06:46:14] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ccf5133
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10907ccf
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4ca07abb149a060e646073b860ccfeaa@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6321ddca
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as3de03838
To: "Office 107" <sip:4141@customer.com>;tag=8D7D9465-CD50C81E
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: af99d365-7ab2b31e-85d1028b@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:46:14] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f68fbe0
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as406b866e
To: "4122" <sip:4122@customer.com>;tag=20E6B604-928E615B
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 1692971-a61359b8-7bc438cf@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:14] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:14] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b17a400
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:46:14]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4109@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bKa6d0b446F288A60B
From: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
To: <sip:4109@customer.com>;tag=as22382717
CSeq: 4 SUBSCRIBE
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="6bf7dcdc", uri="sip:4109@172.30.0.10:5060", response="3db91115a9d142af3bf9689c2a2a10ec", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:15] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ed63e56
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as2d54b824
To: "4124" <sip:4124@customer.com>;tag=4834CBBA-59BBA8E5
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dfbc31d
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as08313b34
To: "4147" <sip:4147@customer.com>;tag=9A4A9A1D-A04807EE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3ebc8d41-ac62f472-8541dfbb@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fdcbd31
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as5b80a547
To: "Office 108" <sip:4128@customer.com>;tag=95EECC8A-82406E13
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 58f32bbe-4088e027-ca4039e8@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7563e7ee
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5f466160
To: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07328dc3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as755a895e
To: "4122" <sip:4122@customer.com>;tag=962E0950-36450E7
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: ec30531d-f8af41a4-bdb82afb@172.30.0.176
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e0d9b14
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as4cc49a90
To: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6321ddca
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as3de03838
To: "Office 107" <sip:4141@customer.com>;tag=8D7D9465-CD50C81E
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: af99d365-7ab2b31e-85d1028b@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f68fbe0
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as406b866e
To: "4122" <sip:4122@customer.com>;tag=20E6B604-928E615B
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 1692971-a61359b8-7bc438cf@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4108@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKa0b5bf8e99AC3D4B
From: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
To: <sip:4108@customer.com>;tag=as23cbc3e8
CSeq: 4 SUBSCRIBE
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="3993d4be", uri="sip:4108@172.30.0.10:5060", response="604a4c1c82965742f5bcc1e7ed483664", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKa0b5bf8e99AC3D4B;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=FC8CA74D-302247DE
To: <sip:4108@customer.com>;tag=as23cbc3e8
Call-ID: 7685f52c-15168611-bf30b852@172.30.0.167
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ed63e56
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as2d54b824
To: "4124" <sip:4124@customer.com>;tag=4834CBBA-59BBA8E5
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dfbc31d
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as08313b34
To: "4147" <sip:4147@customer.com>;tag=9A4A9A1D-A04807EE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3ebc8d41-ac62f472-8541dfbb@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2acf54bc
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fdcbd31
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as5b80a547
To: "Office 108" <sip:4128@customer.com>;tag=95EECC8A-82406E13
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 58f32bbe-4088e027-ca4039e8@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c0e1037
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as22382717
To: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Really destroying SIP dialog '7685f52c-15168611-bf30b852@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.163:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff03f7e
From: <sip:4136@customer.com>;tag=as1938f6a1
To: "4138" <sip:4138@customer.com>;tag=742E799D-D388B26
CSeq: 104 NOTIFY
Call-ID: fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (10 headers 0 lines) ---
[2011-11-03 06:46:15] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:15]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.163:5060
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57acf09c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as035a47b9
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 105901fa1804429c08194de1570d6147@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07328dc3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as755a895e
To: "4122" <sip:4122@customer.com>;tag=962E0950-36450E7
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: ec30531d-f8af41a4-bdb82afb@172.30.0.176
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Really destroying SIP dialog 'fbe2524-c20bf7b1-dbc6ce6a@172.30.0.163' Method: SUBSCRIBE
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.177:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0414ad84
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
CSeq: 110 NOTIFY
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (10 headers 0 lines) ---
[2011-11-03 06:46:15] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:15]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.177:5060
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16c493ae
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as345da678
To: "4140" <sip:4140@customer.com>;tag=8B547CAC-63405B27
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 4f2cbab3-e701fc86-daa75691@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c89e9ca
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as1124fe55
To: "4115" <sip:4115@customer.com>;tag=4E3134F3-82CD450E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: ad81a9a0-c02e6c3b-7b878056@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9f9f08
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as36b775b2
To: "Leon Zhong" <sip:4134@customer.com>;tag=D51BAD8B-9D0E2D48
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 31e9b6e0-cb9bdd39-346616de@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f40698
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as19cd3a67
To: "4123" <sip:4123@customer.com>;tag=A30FF0DF-8006EF6E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: a52898eb-eb76ceda-61e24101@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24ad2f4c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f104a05
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12ec8eed0280cad67a90293e3bd367be@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #9 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4edcbecf
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2acf54bc
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28a56265
Max-Forwards: 70
From: <sip:4171@customer.com>;tag=as2da31670
To: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
Contact: <sip:4171@172.30.0.10:5060>
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4171">
<address uri="sip:4171@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4120@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK16cda37e17A726B
From: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
To: <sip:4120@customer.com>;tag=as203f8f34
CSeq: 4 SUBSCRIBE
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="3179c43f", uri="sip:4120@172.30.0.10:5060", response="a63e04c6816d1f33fb91146d5ec5b7a1", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK16cda37e17A726B;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
To: <sip:4120@customer.com>;tag=as203f8f34
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1855b0e9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1cd45702
To: "Leon Zhong" <sip:4134@customer.com>;tag=237BEF3D-2478B192
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 8722abbe-653b03f-3d5d56ec@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4113@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK13028750B0573E65
From: "4145" <sip:4145@customer.com>;tag=206B3278-C04B246D
To: <sip:4113@customer.com>;tag=as0d4ac688
CSeq: 4 SUBSCRIBE
Call-ID: a4ed9564-8af89fe9-41db734a@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="687043c7", uri="sip:4113@172.30.0.10:5060", response="6979b5406e52b80e0f5770433e5c9918", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK13028750B0573E65;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=206B3278-C04B246D
To: <sip:4113@customer.com>;tag=as0d4ac688
Call-ID: a4ed9564-8af89fe9-41db734a@172.30.0.167
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e0d9b14
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as4cc49a90
To: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b06505b4-1b81c24f-6b7fd06a@172.30.0.197 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6295ms with no response
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6321ddca
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as3de03838
To: "Office 107" <sip:4141@customer.com>;tag=8D7D9465-CD50C81E
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: af99d365-7ab2b31e-85d1028b@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f68fbe0
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as406b866e
To: "4122" <sip:4122@customer.com>;tag=20E6B604-928E615B
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 1692971-a61359b8-7bc438cf@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Really destroying SIP dialog 'b06505b4-1b81c24f-6b7fd06a@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:46:15] Really destroying SIP dialog 'a4ed9564-8af89fe9-41db734a@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4138@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKb533f9e8DB0A6AC3
From: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
To: <sip:4138@customer.com>;tag=as3ee1b4b1
CSeq: 3 SUBSCRIBE
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4168", realm="asterisk", nonce="2fba4568", uri="sip:4138@customer.com:5060", response="9d63c5bb2ce52e06d80ff0eb17ce6a38", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:46:15] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39'
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKb533f9e8DB0A6AC3;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
To: <sip:4138@customer.com>;tag=as3ee1b4b1
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76583eca", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:15] Scheduling destruction of SIP dialog '36f849fc-1ff2a8f7-2bd5446a@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21867acd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as017c026e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e91b4b704344d173a04045939e3b92a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ed63e56
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as2d54b824
To: "4124" <sip:4124@customer.com>;tag=4834CBBA-59BBA8E5
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57cddb12
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as78b92214
To: "4145" <sip:4145@customer.com>;tag=EE058870-AC96B105
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: c1501b8d-a832371e-7927619b@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dfbc31d
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as08313b34
To: "4147" <sip:4147@customer.com>;tag=9A4A9A1D-A04807EE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3ebc8d41-ac62f472-8541dfbb@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fdcbd31
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as5b80a547
To: "Office 108" <sip:4128@customer.com>;tag=95EECC8A-82406E13
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 58f32bbe-4088e027-ca4039e8@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6664aa31
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as50b7453c
To: "4140" <sip:4140@customer.com>;tag=671DAD64-A9BC9A9F
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086e517f
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as239c5cbb
To: "4168" <sip:4168@customer.com>;tag=65549956-704EFFE1
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 98d8ceb7-a724e02a-360d12d5@172.30.0.187
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK39f89515
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as2de2594b
To: "4122" <sip:4122@customer.com>;tag=8E9DF161-47E876A8
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: b25beb44-5ab6b29b-925723c2@172.30.0.176
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b17a400
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68ccc3cb
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as76be7419
To: "Office 107" <sip:4141@customer.com>;tag=147C871-D3F5041A
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: de76fe84-3c752569-af99ea72@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4147@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKcdf3292a598C0B4D
From: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
To: <sip:4147@customer.com>;tag=as31013ea5
CSeq: 3 SUBSCRIBE
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="23508492", uri="sip:4147@customer.com:5060", response="08e8979ac5c25546fc575694c2670887", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:46:15] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106'
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKcdf3292a598C0B4D;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
To: <sip:4147@customer.com>;tag=as31013ea5
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e2beb73", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:46:15] Scheduling destruction of SIP dialog 'b8b4149-dd256cd8-6865693@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0382b7fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5876808c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6fbd360316bd738864155aab6ccded69@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:15] Really destroying SIP dialog 'f02c36fd-a6d8f84-6155c2db@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07328dc3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as755a895e
To: "4122" <sip:4122@customer.com>;tag=962E0950-36450E7
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: ec30531d-f8af41a4-bdb82afb@172.30.0.176
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.183:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK54e0fceb4E16670C
From: "4147" <sip:4147@customer.com>;tag=3F243421-3F10D052
To: <sip:4147@customer.com>
CSeq: 4 REGISTER
Call-ID: c36bdc75-77e4d686-91aca9af@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4147", realm="asterisk", nonce="29154efd", uri="sip:customer.com:5060", response="a3193dba17dab778ea9bd5073fb5f512", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (13 headers 0 lines) ---
[2011-11-03 06:46:15] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:46:15]     -- Unregistered SIP '4147'
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK54e0fceb4E16670C;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=3F243421-3F10D052
To: <sip:4147@customer.com>;tag=as530e21c5
Call-ID: c36bdc75-77e4d686-91aca9af@172.30.0.183
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:15 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:15] set_destination: Parsing <sip:4168@172.30.0.187:5060> for address/port to send to
[2011-11-03 06:46:15] set_destination: set destination to 172.30.0.187:5060
[2011-11-03 06:46:15] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f2051c8
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as329a7cb0
To: "4168" <sip:4168@customer.com>;tag=38F99023-84AA0F76
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: d29db9cf-cfcbe102-eda11f6d@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4168
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:46:15] Scheduling destruction of SIP dialog 'c36bdc75-77e4d686-91aca9af@172.30.0.183' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:46:15] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:46:15] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:46:15] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68c1f546
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as7e04f4e8
To: "4147" <sip:4147@customer.com>;tag=A5843035-597F2446
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 7f99934-1bf475ad-c64fb0fe@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4147
[2011-11-03 06:46:15] set_destination: Parsing <sip:4137@172.30.0.166:5060> for address/port to send to
[2011-11-03 06:46:15] set_destination: set destination to 172.30.0.166:5060
[2011-11-03 06:46:15] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0958c532
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as218c423b
To: "4137" <sip:4137@customer.com>;tag=1420E0B3-E9F68B50
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 5aa222fb-94d11f8-c2938071@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4137
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:15] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:46:15] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:46:15] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK134fe048
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as394157e0
To: "4148" <sip:4148@customer.com>;tag=A3D1C0CE-A5CF4B95
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: a21a407b-937806ea-f7c32d91@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4148
[2011-11-03 06:46:15] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6731560f
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as203f8f34
To: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2acf54bc
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4171@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKf95e9cc444E60EBB
From: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
To: <sip:4171@customer.com>;tag=as2da31670
CSeq: 4 SUBSCRIBE
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="3b25517d", uri="sip:4171@172.30.0.10:5060", response="7a8263e24dc3152a26b4a8d8238da260", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:15] Looking for 4171 in default-local (domain customer.com:5060)
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7563e7ee
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5f466160
To: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c00c1dd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as400a9b8f
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1c08e29e519cfd604e2cfd126020ad30@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4115@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKca688a1a9E4A6EFD
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
CSeq: 4 SUBSCRIBE
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="30b415df", uri="sip:4115@172.30.0.10:5060", response="5201f989dc8738c87c4cc430e54af462", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (16 headers 0 lines) ---
[2011-11-03 06:46:15] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKca688a1a9E4A6EFD;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=4312DE12-FD798A55
To: <sip:4115@customer.com>;tag=as2574012d
Call-ID: 73da2ff0-edbeaf8b-16292a42@172.30.0.165
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f2051c8
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as329a7cb0
To: "4168" <sip:4168@customer.com>;tag=38F99023-84AA0F76
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: d29db9cf-cfcbe102-eda11f6d@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68c1f546
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as7e04f4e8
To: "4147" <sip:4147@customer.com>;tag=A5843035-597F2446
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 7f99934-1bf475ad-c64fb0fe@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ccf5133
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10907ccf
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4ca07abb149a060e646073b860ccfeaa@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0958c532
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as218c423b
To: "4137" <sip:4137@customer.com>;tag=1420E0B3-E9F68B50
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 5aa222fb-94d11f8-c2938071@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK134fe048
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as394157e0
To: "4148" <sip:4148@customer.com>;tag=A3D1C0CE-A5CF4B95
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: a21a407b-937806ea-f7c32d91@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Really destroying SIP dialog '73da2ff0-edbeaf8b-16292a42@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:46:15]
<--- SIP read from UDP:172.30.0.192:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK4941bd643D09BC5B
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>
CSeq: 4 REGISTER
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4148", realm="asterisk", nonce="21bc3c43", uri="sip:customer.com:5060", response="d42f19d93039b7cb677e85609509277e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:15] --- (13 headers 0 lines) ---
[2011-11-03 06:46:15] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:46:15]     -- Unregistered SIP '4148'
[2011-11-03 06:46:15]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK4941bd643D09BC5B;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>;tag=as431ec1e5
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:15 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:15] Scheduling destruction of SIP dialog '9e5886d1-1a3bbf70-bd6245c7@172.30.0.192' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:15] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c0e1037
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as22382717
To: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1f0fab92
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:15] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ecdcc2f
From: <sip:4106@customer.com>;tag=as1fff0b57
To: "4136" <sip:4136@customer.com>;tag=F37A13B7-5AA7ACD0
CSeq: 104 NOTIFY
Call-ID: 39b92ead-60712846-68a8f09f@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (11 headers 0 lines) ---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f2051c8
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as329a7cb0
To: "4168" <sip:4168@customer.com>;tag=38F99023-84AA0F76
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: d29db9cf-cfcbe102-eda11f6d@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68c1f546
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as7e04f4e8
To: "4147" <sip:4147@customer.com>;tag=A5843035-597F2446
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 7f99934-1bf475ad-c64fb0fe@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0958c532
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as218c423b
To: "4137" <sip:4137@customer.com>;tag=1420E0B3-E9F68B50
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 5aa222fb-94d11f8-c2938071@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK134fe048
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as394157e0
To: "4148" <sip:4148@customer.com>;tag=A3D1C0CE-A5CF4B95
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: a21a407b-937806ea-f7c32d91@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.188:5060 --->
SUBSCRIBE sip:4123@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKe931aa3620C6D5A5
From: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
To: <sip:4123@customer.com>;tag=as4a9a0381
CSeq: 4 SUBSCRIBE
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
Contact: <sip:4149@172.30.0.188:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4149", realm="asterisk", nonce="0aff066a", uri="sip:4123@172.30.0.10:5060", response="6eec48e12b23d11fae602e5f6909f326", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (16 headers 0 lines) ---
[2011-11-03 06:46:16] Found peer '4149' for '4149' from 172.30.0.188:5060
[2011-11-03 06:46:16] Looking for 4123 in default-local (domain customer.com:5060)
[2011-11-03 06:46:16]
<--- Transmitting (no NAT) to 172.30.0.188:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKe931aa3620C6D5A5;received=172.30.0.188
From: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
To: <sip:4123@customer.com>;tag=as4a9a0381
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:16] set_destination: Parsing <sip:4149@172.30.0.188:5060> for address/port to send to
[2011-11-03 06:46:16] set_destination: set destination to 172.30.0.188:5060
[2011-11-03 06:46:16] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47727bf1
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as4a9a0381
To: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4edcbecf
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
CSeq: 104 NOTIFY
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (11 headers 0 lines) ---
[2011-11-03 06:46:16] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.192:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK4941bd643D09BC5B
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>
CSeq: 4 REGISTER
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4148", realm="asterisk", nonce="21bc3c43", uri="sip:customer.com:5060", response="d42f19d93039b7cb677e85609509277e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (13 headers 0 lines) ---
[2011-11-03 06:46:16] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:46:16]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK4941bd643D09BC5B;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>;tag=as431ec1e5
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:16 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:16] Scheduling destruction of SIP dialog '9e5886d1-1a3bbf70-bd6245c7@172.30.0.192' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57acf09c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as035a47b9
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 105901fa1804429c08194de1570d6147@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:16] Retransmitting #6 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK309e3095
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as63503940
To: "4108" <sip:4108@customer.com>;tag=1D22334E-CA8574E9
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: bc846304-17b3879f-6fd7fdba@172.30.0.196
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47727bf1
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as4a9a0381
To: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4120@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKf7ff32b6CC22FFC9
From: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
To: <sip:4120@customer.com>;tag=as6f074e76
CSeq: 4 SUBSCRIBE
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="7e7166ca", uri="sip:4120@172.30.0.10:5060", response="978b3321e0b34327aff04e496f181cf0", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (16 headers 0 lines) ---
[2011-11-03 06:46:16] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:46:16] Looking for 4120 in default-local (domain customer.com:5060)
[2011-11-03 06:46:16]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKf7ff32b6CC22FFC9;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
To: <sip:4120@customer.com>;tag=as6f074e76
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:16] set_destination: Parsing <sip:4171@172.30.0.165:5060> for address/port to send to
[2011-11-03 06:46:16] set_destination: set destination to 172.30.0.165:5060
[2011-11-03 06:46:16] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59081b1c
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as6f074e76
To: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e0d9b14
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as4cc49a90
To: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6321ddca
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as3de03838
To: "Office 107" <sip:4141@customer.com>;tag=8D7D9465-CD50C81E
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: af99d365-7ab2b31e-85d1028b@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #6 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK70fde51f
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as1105be02
To: "Office 108" <sip:4128@customer.com>;tag=E12954C5-5C6E56
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4d5c1f87-46dbba48-db35e6a1@172.30.0.182
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f68fbe0
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as406b866e
To: "4122" <sip:4122@customer.com>;tag=20E6B604-928E615B
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 1692971-a61359b8-7bc438cf@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24ad2f4c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f104a05
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12ec8eed0280cad67a90293e3bd367be@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:16] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission bc846304-17b3879f-6fd7fdba@172.30.0.196 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6398ms with no response
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ed63e56
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as2d54b824
To: "4124" <sip:4124@customer.com>;tag=4834CBBA-59BBA8E5
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dfbc31d
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as08313b34
To: "4147" <sip:4147@customer.com>;tag=9A4A9A1D-A04807EE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3ebc8d41-ac62f472-8541dfbb@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fdcbd31
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as5b80a547
To: "Office 108" <sip:4128@customer.com>;tag=95EECC8A-82406E13
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 58f32bbe-4088e027-ca4039e8@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Really destroying SIP dialog 'bc846304-17b3879f-6fd7fdba@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4116@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK56c7972a54E30FD1
From: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
To: <sip:4116@customer.com>;tag=as1b71ddfa
CSeq: 4 SUBSCRIBE
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="5b15b1cd", uri="sip:4116@172.30.0.10:5060", response="73a772a79d0073798cc1da7857bf7c1d", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (16 headers 0 lines) ---
[2011-11-03 06:46:16] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:46:16] Looking for 4116 in default-local (domain customer.com:5060)
[2011-11-03 06:46:16]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK56c7972a54E30FD1;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
To: <sip:4116@customer.com>;tag=as1b71ddfa
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Content-Length: 0


<------------>
[2011-11-03 06:46:16] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:16] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:16] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK638d0430
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as1b71ddfa
To: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 4d5c1f87-46dbba48-db35e6a1@172.30.0.182 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6418ms with no response
[2011-11-03 06:46:16] Retransmitting #5 (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50859a83
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as43fff8b0
To: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.163:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f882d2f
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c079792
To: <sip:4138@172.30.0.163:5060>;tag=C3D5A2D8-86404815
CSeq: 102 OPTIONS
Call-ID: 1eb35a210a1486d8109bf10428ed556f@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (9 headers 0 lines) ---
[2011-11-03 06:46:16] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59081b1c
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as6f074e76
To: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07328dc3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as755a895e
To: "4122" <sip:4122@customer.com>;tag=962E0950-36450E7
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: ec30531d-f8af41a4-bdb82afb@172.30.0.176
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #8 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK755a9481
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #6 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e66be5d
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as5f5dbb11
To: "4116" <sip:4116@customer.com>;tag=AD74E8F2-B480803B
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f2051c8
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as329a7cb0
To: "4168" <sip:4168@customer.com>;tag=38F99023-84AA0F76
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: d29db9cf-cfcbe102-eda11f6d@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47727bf1
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as4a9a0381
To: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.188:5060 --->
SUBSCRIBE sip:4123@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKe931aa3620C6D5A5
From: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
To: <sip:4123@customer.com>;tag=as4a9a0381
CSeq: 4 SUBSCRIBE
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
Contact: <sip:4149@172.30.0.188:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4149", realm="asterisk", nonce="0aff066a", uri="sip:4123@172.30.0.10:5060", response="6eec48e12b23d11fae602e5f6909f326", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (16 headers 0 lines) ---
[2011-11-03 06:46:16] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:16] Looking for 4123 in default-local (domain customer.com:5060)
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68c1f546
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as7e04f4e8
To: "4147" <sip:4147@customer.com>;tag=A5843035-597F2446
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 7f99934-1bf475ad-c64fb0fe@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0958c532
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as218c423b
To: "4137" <sip:4137@customer.com>;tag=1420E0B3-E9F68B50
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 5aa222fb-94d11f8-c2938071@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK134fe048
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as394157e0
To: "4148" <sip:4148@customer.com>;tag=A3D1C0CE-A5CF4B95
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: a21a407b-937806ea-f7c32d91@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21867acd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as017c026e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e91b4b704344d173a04045939e3b92a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:16] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK638d0430
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as1b71ddfa
To: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2acf54bc
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 932bc7b7-ad540ef8-83c8d3d1@172.30.0.199 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6408ms with no response
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b17a400
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Really destroying SIP dialog '932bc7b7-ad540ef8-83c8d3d1@172.30.0.199' Method: SUBSCRIBE
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: "4136" <sip:4136@172.30.0.172:5060>;tag=D4D85BE7-502B5900
CSeq: 102 OPTIONS
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
Contact: <sip:4136@172.30.0.172:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (14 headers 0 lines) ---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0382b7fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5876808c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6fbd360316bd738864155aab6ccded69@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:16] Really destroying SIP dialog '6fbd360316bd738864155aab6ccded69@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:16] Retransmitting #7 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff6cbdd
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as284d87e4
To: "4118" <sip:4118@customer.com>;tag=E9AFD17D-87EBC8D2
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 28129ff1-31250646-5c60a25b@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59081b1c
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as6f074e76
To: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK118d53f7
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6913496f
To: "4142" <sip:4142@172.30.0.198:5060>;tag=345B318F-BC0E3206
CSeq: 102 OPTIONS
Call-ID: 56b4d35a79ed333b65582e2b773a21f2@172.30.0.10:5060
Contact: <sip:4142@172.30.0.198:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (14 headers 0 lines) ---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.183:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK54e0fceb4E16670C
From: "4147" <sip:4147@customer.com>;tag=3F243421-3F10D052
To: <sip:4147@customer.com>
CSeq: 4 REGISTER
Call-ID: c36bdc75-77e4d686-91aca9af@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4147", realm="asterisk", nonce="29154efd", uri="sip:customer.com:5060", response="a3193dba17dab778ea9bd5073fb5f512", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (13 headers 0 lines) ---
[2011-11-03 06:46:16] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:46:16]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK54e0fceb4E16670C;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=3F243421-3F10D052
To: <sip:4147@customer.com>;tag=as530e21c5
Call-ID: c36bdc75-77e4d686-91aca9af@172.30.0.183
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:16 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:16] Scheduling destruction of SIP dialog 'c36bdc75-77e4d686-91aca9af@172.30.0.183' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:16] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK638d0430
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as1b71ddfa
To: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:46:16] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:46:16] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5520b03f
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.192:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK371c1d9a
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as048c45eb
To: <sip:4148@172.30.0.192:5060>;tag=B4601510-7CE91067
CSeq: 102 OPTIONS
Call-ID: 54fcd2432666720340cb6f795fdc2fc8@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (9 headers 0 lines) ---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4148@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK14e423c53DC32556
From: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
To: <sip:4148@customer.com>;tag=as4b9f0e07
CSeq: 4 SUBSCRIBE
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4147", realm="asterisk", nonce="362eb861", uri="sip:4148@172.30.0.10:5060", response="cc73447d40b2467fb9ce3f85a26e7446", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (16 headers 0 lines) ---
[2011-11-03 06:46:16] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.
[2011-11-03 06:46:16]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK14e423c53DC32556;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
To: <sip:4148@customer.com>;tag=as4b9f0e07
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4120@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKf7ff32b6CC22FFC9
From: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
To: <sip:4120@customer.com>;tag=as6f074e76
CSeq: 4 SUBSCRIBE
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="7e7166ca", uri="sip:4120@172.30.0.10:5060", response="978b3321e0b34327aff04e496f181cf0", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (16 headers 0 lines) ---
[2011-11-03 06:46:16] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:16] Looking for 4120 in default-local (domain customer.com:5060)
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c00c1dd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as400a9b8f
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1c08e29e519cfd604e2cfd126020ad30@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:16] Really destroying SIP dialog '1c08e29e519cfd604e2cfd126020ad30@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:16] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5520b03f
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47727bf1
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as4a9a0381
To: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ccf5133
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10907ccf
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4ca07abb149a060e646073b860ccfeaa@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:16] Really destroying SIP dialog '4ca07abb149a060e646073b860ccfeaa@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:16] Retransmitting #5 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16c493ae
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as345da678
To: "4140" <sip:4140@customer.com>;tag=8B547CAC-63405B27
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 4f2cbab3-e701fc86-daa75691@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #5 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c89e9ca
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as1124fe55
To: "4115" <sip:4115@customer.com>;tag=4E3134F3-82CD450E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: ad81a9a0-c02e6c3b-7b878056@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9f9f08
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as36b775b2
To: "Leon Zhong" <sip:4134@customer.com>;tag=D51BAD8B-9D0E2D48
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 31e9b6e0-cb9bdd39-346616de@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK077564f8
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f6f12ce
To: <sip:4171@172.30.0.165:5060>;tag=CBC28D58-9D668113
CSeq: 102 OPTIONS
Call-ID: 460c26e35e974dac4c5bc63c43be7ff2@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (9 headers 0 lines) ---
[2011-11-03 06:46:16] Retransmitting #5 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f40698
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as19cd3a67
To: "4123" <sip:4123@customer.com>;tag=A30FF0DF-8006EF6E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: a52898eb-eb76ceda-61e24101@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1855b0e9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1cd45702
To: "Leon Zhong" <sip:4134@customer.com>;tag=237BEF3D-2478B192
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 8722abbe-653b03f-3d5d56ec@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:16]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4116@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK56c7972a54E30FD1
From: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
To: <sip:4116@customer.com>;tag=as1b71ddfa
CSeq: 4 SUBSCRIBE
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="5b15b1cd", uri="sip:4116@172.30.0.10:5060", response="73a772a79d0073798cc1da7857bf7c1d", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:16] --- (16 headers 0 lines) ---
[2011-11-03 06:46:16] Ignoring this SUBSCRIBE request
[2011-11-03 06:46:16] Looking for 4116 in default-local (domain customer.com:5060)
[2011-11-03 06:46:17] Retransmitting #10 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK349b135e
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as29c5093b
To: "4136" <sip:4136@customer.com>;tag=AF9D5AC3-4C80CDC
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: 697be950-161e5129-5d1a37c2@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59081b1c
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as6f074e76
To: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5520b03f
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
CSeq: 105 NOTIFY
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (10 headers 0 lines) ---
[2011-11-03 06:46:17] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:17]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.165:5060
[2011-11-03 06:46:17] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK638d0430
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as1b71ddfa
To: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Really destroying SIP dialog 'dc8827b9-82fb4408-db62ad83@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28a56265
From: <sip:4171@customer.com>;tag=as2da31670
To: "4173" <sip:4173@customer.com>;tag=497ECB75-910AE34
CSeq: 104 NOTIFY
Call-ID: 69de6be3-ca1cff92-22b56d79@172.30.0.178
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (10 headers 0 lines) ---
[2011-11-03 06:46:17] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:17]     -- Incoming call: Got SIP response 403 "Forbidden" back from 172.30.0.178:5060
[2011-11-03 06:46:17] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57acf09c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as035a47b9
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 105901fa1804429c08194de1570d6147@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:17] Really destroying SIP dialog '69de6be3-ca1cff92-22b56d79@172.30.0.178' Method: SUBSCRIBE
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
CSeq: 105 NOTIFY
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (10 headers 0 lines) ---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f2051c8
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as329a7cb0
To: "4168" <sip:4168@customer.com>;tag=38F99023-84AA0F76
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: d29db9cf-cfcbe102-eda11f6d@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68c1f546
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as7e04f4e8
To: "4147" <sip:4147@customer.com>;tag=A5843035-597F2446
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 7f99934-1bf475ad-c64fb0fe@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0958c532
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as218c423b
To: "4137" <sip:4137@customer.com>;tag=1420E0B3-E9F68B50
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 5aa222fb-94d11f8-c2938071@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK134fe048
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as394157e0
To: "4148" <sip:4148@customer.com>;tag=A3D1C0CE-A5CF4B95
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: a21a407b-937806ea-f7c32d91@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.192:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK4941bd643D09BC5B
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>
CSeq: 4 REGISTER
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>;expires=0
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4148", realm="asterisk", nonce="21bc3c43", uri="sip:customer.com:5060", response="d42f19d93039b7cb677e85609509277e", algorithm=MD5
Max-Forwards: 70
Expires: 0
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (13 headers 0 lines) ---
[2011-11-03 06:46:17] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:46:17]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK4941bd643D09BC5B;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=F0B00C85-7F9B2BC4
To: <sip:4148@customer.com>;tag=as431ec1e5
Call-ID: 9e5886d1-1a3bbf70-bd6245c7@172.30.0.192
CSeq: 4 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 0
Date: Thu, 03 Nov 2011 13:46:17 GMT
Content-Length: 0


<------------>
[2011-11-03 06:46:17] Scheduling destruction of SIP dialog '9e5886d1-1a3bbf70-bd6245c7@172.30.0.192' in 32000 ms (Method: REGISTER)
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24ad2f4c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f104a05
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12ec8eed0280cad67a90293e3bd367be@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:17] Really destroying SIP dialog '12ec8eed0280cad67a90293e3bd367be@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
CSeq: 105 NOTIFY
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (10 headers 0 lines) ---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7563e7ee
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5f466160
To: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 697be950-161e5129-5d1a37c2@172.30.0.172 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31929ms with no response
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: "4136" <sip:4136@172.30.0.172:5060>;tag=D4D85BE7-502B5900
CSeq: 102 OPTIONS
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
Contact: <sip:4136@172.30.0.172:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (14 headers 0 lines) ---
[2011-11-03 06:46:17] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5520b03f
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.187:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c00c1dd
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as400a9b8f
To: <sip:4168@172.30.0.187:5060>;tag=6F71CFB8-765BF513
CSeq: 102 OPTIONS
Call-ID: 1c08e29e519cfd604e2cfd126020ad30@172.30.0.10:5060
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (9 headers 0 lines) ---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK21867acd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as017c026e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e91b4b704344d173a04045939e3b92a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:17] Really destroying SIP dialog '3e91b4b704344d173a04045939e3b92a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afd1e8b
From: <sip:4130@customer.com>;tag=as1443a841
To: "Office 113" <sip:4171@customer.com>;tag=FAA57BA2-7DCD8725
CSeq: 105 NOTIFY
Call-ID: dc8827b9-82fb4408-db62ad83@172.30.0.165
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (10 headers 0 lines) ---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c0e1037
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as22382717
To: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #7 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0975328c
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as728e16e0
To: "4136" <sip:4136@customer.com>;tag=19F02E68-92FA0841
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: cc56bb5f-1e5b7278-c0f81851@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0975328c
From: <sip:4116@customer.com>;tag=as728e16e0
To: "4136" <sip:4136@customer.com>;tag=19F02E68-92FA0841
CSeq: 104 NOTIFY
Call-ID: cc56bb5f-1e5b7278-c0f81851@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (11 headers 0 lines) ---
[2011-11-03 06:46:17] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6daedae7
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as496ef9a5
To: "4136" <sip:4136@172.30.0.172:5060>;tag=D4D85BE7-502B5900
CSeq: 102 OPTIONS
Call-ID: 6d0092414896b71c2a8c45716fe8c475@172.30.0.10:5060
Contact: <sip:4136@172.30.0.172:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (14 headers 0 lines) ---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47727bf1
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as4a9a0381
To: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
CSeq: 105 NOTIFY
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (11 headers 0 lines) ---
[2011-11-03 06:46:17] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:17] Retransmitting #10 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7a7c8ee1
Max-Forwards: 70
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
Contact: <sip:4118@172.30.0.10:5060>
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4118">
<address uri="sip:4118@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7a7c8ee1
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
CSeq: 104 NOTIFY
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (11 headers 0 lines) ---
[2011-11-03 06:46:17] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:17] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:46:17] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:46:17] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e175806
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5fc896fe
To: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
CSeq: 105 NOTIFY
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (11 headers 0 lines) ---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59081b1c
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as6f074e76
To: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e0d9b14
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as4cc49a90
To: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1f0fab92
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
CSeq: 104 NOTIFY
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (11 headers 0 lines) ---
[2011-11-03 06:46:17] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6321ddca
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as3de03838
To: "Office 107" <sip:4141@customer.com>;tag=8D7D9465-CD50C81E
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: af99d365-7ab2b31e-85d1028b@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f68fbe0
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as406b866e
To: "4122" <sip:4122@customer.com>;tag=20E6B604-928E615B
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 1692971-a61359b8-7bc438cf@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK638d0430
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as1b71ddfa
To: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ed63e56
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as2d54b824
To: "4124" <sip:4124@customer.com>;tag=4834CBBA-59BBA8E5
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e175806
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5fc896fe
To: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dfbc31d
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as08313b34
To: "4147" <sip:4147@customer.com>;tag=9A4A9A1D-A04807EE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3ebc8d41-ac62f472-8541dfbb@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17] Retransmitting #5 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fdcbd31
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as5b80a547
To: "Office 108" <sip:4128@customer.com>;tag=95EECC8A-82406E13
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 58f32bbe-4088e027-ca4039e8@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:17]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
CSeq: 105 NOTIFY
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:17] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
CSeq: 105 NOTIFY
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #5 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07328dc3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as755a895e
To: "4122" <sip:4122@customer.com>;tag=962E0950-36450E7
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: ec30531d-f8af41a4-bdb82afb@172.30.0.176
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] set_destination: Parsing <sip:4123@172.30.0.184:5060> for address/port to send to
[2011-11-03 06:46:18] set_destination: set destination to 172.30.0.184:5060
[2011-11-03 06:46:18] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK105fabbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2c27a83d
To: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
CSeq: 104 NOTIFY
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:18] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e175806
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5fc896fe
To: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2acf54bc
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57acf09c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as035a47b9
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 105901fa1804429c08194de1570d6147@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:18] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4147' is now UNREACHABLE!  Last qualify: 61
[2011-11-03 06:46:18] Really destroying SIP dialog '105901fa1804429c08194de1570d6147@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:18] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5520b03f
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b17a400
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
CSeq: 104 NOTIFY
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:18] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK105fabbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2c27a83d
To: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
CSeq: 104 NOTIFY
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
CSeq: 104 NOTIFY
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
CSeq: 105 NOTIFY
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e175806
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5fc896fe
To: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4edcbecf
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
CSeq: 104 NOTIFY
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK105fabbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2c27a83d
To: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
CSeq: 104 NOTIFY
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #10 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b17a400
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
CSeq: 104 NOTIFY
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1f0fab92
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
CSeq: 104 NOTIFY
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
CSeq: 104 NOTIFY
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #6 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57cddb12
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as78b92214
To: "4145" <sip:4145@customer.com>;tag=EE058870-AC96B105
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: c1501b8d-a832371e-7927619b@172.30.0.167
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #6 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6664aa31
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as50b7453c
To: "4140" <sip:4140@customer.com>;tag=671DAD64-A9BC9A9F
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.175
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #6 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086e517f
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as239c5cbb
To: "4168" <sip:4168@customer.com>;tag=65549956-704EFFE1
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 98d8ceb7-a724e02a-360d12d5@172.30.0.187
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #6 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK39f89515
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as2de2594b
To: "4122" <sip:4122@customer.com>;tag=8E9DF161-47E876A8
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: b25beb44-5ab6b29b-925723c2@172.30.0.176
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK755a9481
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
CSeq: 107 NOTIFY
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:18] Retransmitting #6 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68ccc3cb
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as76be7419
To: "Office 107" <sip:4141@customer.com>;tag=147C871-D3F5041A
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: de76fe84-3c752569-af99ea72@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c1501b8d-a832371e-7927619b@172.30.0.167 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6380ms with no response
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b17a400
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
CSeq: 104 NOTIFY
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e9d3782e-2da61a79-58cc69bc@172.30.0.175 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6418ms with no response
[2011-11-03 06:46:18] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission de76fe84-3c752569-af99ea72@172.30.0.181 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6388ms with no response
[2011-11-03 06:46:18] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 98d8ceb7-a724e02a-360d12d5@172.30.0.187 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6413ms with no response
[2011-11-03 06:46:18] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b25beb44-5ab6b29b-925723c2@172.30.0.176 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6406ms with no response
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4e2cc0e3
From: <sip:4128@customer.com>;tag=as45a898f4
To: "4136" <sip:4136@customer.com>;tag=429BFF4E-41B0EDA7
CSeq: 105 NOTIFY
Call-ID: d4517688-dd90e861-e0c958fa@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #5 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f2051c8
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as329a7cb0
To: "4168" <sip:4168@customer.com>;tag=38F99023-84AA0F76
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: d29db9cf-cfcbe102-eda11f6d@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68c1f546
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as7e04f4e8
To: "4147" <sip:4147@customer.com>;tag=A5843035-597F2446
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 7f99934-1bf475ad-c64fb0fe@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #5 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0958c532
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as218c423b
To: "4137" <sip:4137@customer.com>;tag=1420E0B3-E9F68B50
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 5aa222fb-94d11f8-c2938071@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK134fe048
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as394157e0
To: "4148" <sip:4148@customer.com>;tag=A3D1C0CE-A5CF4B95
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: a21a407b-937806ea-f7c32d91@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #6 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6731560f
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as203f8f34
To: "Office 107" <sip:4141@customer.com>;tag=9C4120FD-89FBDE96
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 38a4be17-cb8f6e38-3efd882d@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK349b135e
From: <sip:4142@customer.com>;tag=as29c5093b
To: "4136" <sip:4136@customer.com>;tag=AF9D5AC3-4C80CDC
CSeq: 104 NOTIFY
Call-ID: 697be950-161e5129-5d1a37c2@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK105fabbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2c27a83d
To: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Really destroying SIP dialog '38a4be17-cb8f6e38-3efd882d@172.30.0.181' Method: SUBSCRIBE
[2011-11-03 06:46:18]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0975328c
From: <sip:4116@customer.com>;tag=as728e16e0
To: "4136" <sip:4136@customer.com>;tag=19F02E68-92FA0841
CSeq: 104 NOTIFY
Call-ID: cc56bb5f-1e5b7278-c0f81851@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:18] --- (11 headers 0 lines) ---
[2011-11-03 06:46:18] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:18] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:19]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7a7c8ee1
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
CSeq: 104 NOTIFY
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:19] --- (11 headers 0 lines) ---
[2011-11-03 06:46:19]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28f84200
From: <sip:4123@customer.com>;tag=as29ff8a73
To: "4136" <sip:4136@customer.com>;tag=A7B5F27B-6F1B8E94
CSeq: 104 NOTIFY
Call-ID: 136610d6-435a052f-3a1ed848@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:19] --- (11 headers 0 lines) ---
[2011-11-03 06:46:19] Retransmitting #5 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47727bf1
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as4a9a0381
To: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:19] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e175806
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5fc896fe
To: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:19] Retransmitting #5 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59081b1c
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as6f074e76
To: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:19] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK638d0430
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as1b71ddfa
To: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:19] Retransmitting #6 (no NAT) to 172.30.0.170:5060:
NOTIFY sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50859a83
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as43fff8b0
To: "Office 105" <sip:4131@customer.com>;tag=72613BED-BB445850
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c6598b22-5d64568d-28d073f0@172.30.0.170
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4131@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:19] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c6598b22-5d64568d-28d073f0@172.30.0.170 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2011-11-03 06:46:19] Really destroying SIP dialog 'c6598b22-5d64568d-28d073f0@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:46:19] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK105fabbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2c27a83d
To: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:19] Really destroying SIP dialog '378e3b42-e0b6ca05-40accd3c@172.30.0.189' Method: REGISTER
[2011-11-03 06:46:19] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5520b03f
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:46:20] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:46:20] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK108555bd
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as40f08f92
To: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #6 (no NAT) to 172.30.0.175:5060:
NOTIFY sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16c493ae
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as345da678
To: "4140" <sip:4140@customer.com>;tag=8B547CAC-63405B27
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 4f2cbab3-e701fc86-daa75691@172.30.0.175
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4140@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #6 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c89e9ca
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as1124fe55
To: "4115" <sip:4115@customer.com>;tag=4E3134F3-82CD450E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: ad81a9a0-c02e6c3b-7b878056@172.30.0.197
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9f9f08
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as36b775b2
To: "Leon Zhong" <sip:4134@customer.com>;tag=D51BAD8B-9D0E2D48
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: 31e9b6e0-cb9bdd39-346616de@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #6 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f40698
Max-Forwards: 70
From: <sip:4128@customer.com>;tag=as19cd3a67
To: "4123" <sip:4123@customer.com>;tag=A30FF0DF-8006EF6E
Contact: <sip:4128@172.30.0.10:5060>
Call-ID: a52898eb-eb76ceda-61e24101@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4128">
<address uri="sip:4128@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 4f2cbab3-e701fc86-daa75691@172.30.0.175 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6414ms with no response
[2011-11-03 06:46:20] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ad81a9a0-c02e6c3b-7b878056@172.30.0.197 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6414ms with no response
[2011-11-03 06:46:20] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 31e9b6e0-cb9bdd39-346616de@172.30.0.179 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6411ms with no response
[2011-11-03 06:46:20] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1855b0e9
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as1cd45702
To: "Leon Zhong" <sip:4134@customer.com>;tag=237BEF3D-2478B192
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 8722abbe-653b03f-3d5d56ec@172.30.0.179
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a52898eb-eb76ceda-61e24101@172.30.0.184 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6409ms with no response
[2011-11-03 06:46:20] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK108555bd
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as40f08f92
To: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 8722abbe-653b03f-3d5d56ec@172.30.0.179 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6412ms with no response
[2011-11-03 06:46:20] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK108555bd
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as40f08f92
To: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #6 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7563e7ee
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5f466160
To: "4135" <sip:4135@customer.com>;tag=43345021-FAB9A9A6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: bc87819c-cb1e1045-a5c35ffa@172.30.0.177
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #8 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff6cbdd
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as284d87e4
To: "4118" <sip:4118@customer.com>;tag=E9AFD17D-87EBC8D2
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 28129ff1-31250646-5c60a25b@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission bc87819c-cb1e1045-a5c35ffa@172.30.0.177 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2011-11-03 06:46:20] Really destroying SIP dialog 'bc87819c-cb1e1045-a5c35ffa@172.30.0.177' Method: SUBSCRIBE
[2011-11-03 06:46:20] Retransmitting #6 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c0e1037
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as22382717
To: "4138" <sip:4138@customer.com>;tag=BE0C9BFC-2704B829
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 623b63cb-a9e846dc-18385b89@172.30.0.163
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK108555bd
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as40f08f92
To: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 623b63cb-a9e846dc-18385b89@172.30.0.163 for seqno 107 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6391ms with no response
[2011-11-03 06:46:20] Really destroying SIP dialog '623b63cb-a9e846dc-18385b89@172.30.0.163' Method: SUBSCRIBE
[2011-11-03 06:46:20] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:20] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e175806
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5fc896fe
To: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #6 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e0d9b14
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as4cc49a90
To: "4115" <sip:4115@customer.com>;tag=59A2AE9-40003E04
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 191ae026-424cadc1-eaf44cdc@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #6 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6321ddca
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as3de03838
To: "Office 107" <sip:4141@customer.com>;tag=8D7D9465-CD50C81E
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: af99d365-7ab2b31e-85d1028b@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #6 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6f68fbe0
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as406b866e
To: "4122" <sip:4122@customer.com>;tag=20E6B604-928E615B
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 1692971-a61359b8-7bc438cf@172.30.0.176
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ed63e56
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as2d54b824
To: "4124" <sip:4124@customer.com>;tag=4834CBBA-59BBA8E5
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dfbc31d
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as08313b34
To: "4147" <sip:4147@customer.com>;tag=9A4A9A1D-A04807EE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3ebc8d41-ac62f472-8541dfbb@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #6 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fdcbd31
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as5b80a547
To: "Office 108" <sip:4128@customer.com>;tag=95EECC8A-82406E13
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 58f32bbe-4088e027-ca4039e8@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 191ae026-424cadc1-eaf44cdc@172.30.0.197 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6423ms with no response
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission af99d365-7ab2b31e-85d1028b@172.30.0.181 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6396ms with no response
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 1692971-a61359b8-7bc438cf@172.30.0.176 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6387ms with no response
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 58f32bbe-4088e027-ca4039e8@172.30.0.182 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6416ms with no response
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 81e5b41d-f0e5cc00-d51caa9b@172.30.0.164 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6434ms with no response
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3ebc8d41-ac62f472-8541dfbb@172.30.0.183 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6429ms with no response
[2011-11-03 06:46:21] Retransmitting #6 (no NAT) to 172.30.0.176:5060:
NOTIFY sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07328dc3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as755a895e
To: "4122" <sip:4122@customer.com>;tag=962E0950-36450E7
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: ec30531d-f8af41a4-bdb82afb@172.30.0.176
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4122@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission cf08829-f486811a-8885c823@172.30.0.183 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6271ms with no response
[2011-11-03 06:46:21] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:46:21] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:46:21] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5ded4d
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Really destroying SIP dialog '9e5886d1-1a3bbf70-bd6245c7@172.30.0.184' Method: REGISTER
[2011-11-03 06:46:21] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ec30531d-f8af41a4-bdb82afb@172.30.0.176 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6422ms with no response
[2011-11-03 06:46:21] Retransmitting #5 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK105fabbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2c27a83d
To: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5ded4d
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK108555bd
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as40f08f92
To: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Really destroying SIP dialog '64656865-be56e5ce-ee8e073@172.30.0.163' Method: REGISTER
[2011-11-03 06:46:21] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5ded4d
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45414ebd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a2738ce
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 700f69796366f16d3717ece77a7687ba@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:21] set_destination: Parsing <sip:4168@172.30.0.187:5060> for address/port to send to
[2011-11-03 06:46:21] set_destination: set destination to 172.30.0.187:5060
[2011-11-03 06:46:21] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f5726eb
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3ee1b4b1
To: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:21] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0aaee423
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7aac0f75
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e9efc9066d9e6fa5f310e0856f681f7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:21] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72c611ce
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50e27b2a
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0be0c0dc001e1ad1173a40f548c90d27@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5ded4d
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] set_destination: Parsing <sip:4171@172.30.0.165:5060> for address/port to send to
[2011-11-03 06:46:22] set_destination: set destination to 172.30.0.165:5060
[2011-11-03 06:46:22] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK63b4083f
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as31013ea5
To: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #6 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f2051c8
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as329a7cb0
To: "4168" <sip:4168@customer.com>;tag=38F99023-84AA0F76
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: d29db9cf-cfcbe102-eda11f6d@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68c1f546
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as7e04f4e8
To: "4147" <sip:4147@customer.com>;tag=A5843035-597F2446
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 7f99934-1bf475ad-c64fb0fe@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f5726eb
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3ee1b4b1
To: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #6 (no NAT) to 172.30.0.166:5060:
NOTIFY sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0958c532
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as218c423b
To: "4137" <sip:4137@customer.com>;tag=1420E0B3-E9F68B50
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 5aa222fb-94d11f8-c2938071@172.30.0.166
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4137@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK134fe048
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as394157e0
To: "4148" <sip:4148@customer.com>;tag=A3D1C0CE-A5CF4B95
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: a21a407b-937806ea-f7c32d91@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission d29db9cf-cfcbe102-eda11f6d@172.30.0.187 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6404ms with no response
[2011-11-03 06:46:22] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 7f99934-1bf475ad-c64fb0fe@172.30.0.183 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6387ms with no response
[2011-11-03 06:46:22] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 5aa222fb-94d11f8-c2938071@172.30.0.166 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6382ms with no response
[2011-11-03 06:46:22] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a21a407b-937806ea-f7c32d91@172.30.0.192 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6428ms with no response
[2011-11-03 06:46:22] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37db833a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as62eb1ab4
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4af1034b2be7d5a14ba394147e5ecd90@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK63b4083f
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as31013ea5
To: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f5726eb
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3ee1b4b1
To: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK63b4083f
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as31013ea5
To: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #6 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47727bf1
Max-Forwards: 70
From: <sip:4123@customer.com>;tag=as4a9a0381
To: "4149" <sip:4149@customer.com>;tag=E6B78517-5F2936DE
Contact: <sip:4123@172.30.0.10:5060>
Call-ID: 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4123">
<address uri="sip:4123@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3a4a0277-34a32d3e-ce93ab6d@172.30.0.188 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6409ms with no response
[2011-11-03 06:46:22] Really destroying SIP dialog '9e5886d1-1a3bbf70-bd6245c7@172.30.0.180' Method: REGISTER
[2011-11-03 06:46:22] Really destroying SIP dialog '3a4a0277-34a32d3e-ce93ab6d@172.30.0.188' Method: SUBSCRIBE
[2011-11-03 06:46:22] Retransmitting #11 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #6 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59081b1c
Max-Forwards: 70
From: <sip:4120@customer.com>;tag=as6f074e76
To: "Office 113" <sip:4171@customer.com>;tag=F2E5A902-F07BA005
Contact: <sip:4120@172.30.0.10:5060>
Call-ID: 57d0a5f6-6d5ab409-9e569698@172.30.0.165
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4120">
<address uri="sip:4120@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f5726eb
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3ee1b4b1
To: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f355269
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0086ce3d
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3b5f80e86a42c54234f99a834c170906@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK638d0430
Max-Forwards: 70
From: <sip:4116@customer.com>;tag=as1b71ddfa
To: "4173" <sip:4173@customer.com>;tag=50FDB04E-A5671715
Contact: <sip:4116@172.30.0.10:5060>
Call-ID: 549cc804-ea7523fb-947f666a@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4116">
<address uri="sip:4116@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 57d0a5f6-6d5ab409-9e569698@172.30.0.165 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6415ms with no response
[2011-11-03 06:46:22] Really destroying SIP dialog '57d0a5f6-6d5ab409-9e569698@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:46:22] Really destroying SIP dialog '4d5c1f87-46dbba48-db35e6a1@172.30.0.182' Method: SUBSCRIBE
[2011-11-03 06:46:22] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 549cc804-ea7523fb-947f666a@172.30.0.178 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6407ms with no response
[2011-11-03 06:46:22] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK63b4083f
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as31013ea5
To: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Really destroying SIP dialog '549cc804-ea7523fb-947f666a@172.30.0.178' Method: SUBSCRIBE
[2011-11-03 06:46:22] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5ded4d
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Reliably Transmitting (no NAT) to 172.30.0.198:5060:
OPTIONS sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57c8072a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ac1c847
To: <sip:4142@172.30.0.198:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4e007a445f5cb571765a94c466bcb0cc@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533a3de2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as611f31d0
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6c2e6a7f3781f85b17b1c5a05d449c22@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22]
<--- SIP read from UDP:172.30.0.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK57c8072a
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ac1c847
To: "4142" <sip:4142@172.30.0.198:5060>;tag=79C9F530-9DAB7503
CSeq: 102 OPTIONS
Call-ID: 4e007a445f5cb571765a94c466bcb0cc@172.30.0.10:5060
Contact: <sip:4142@172.30.0.198:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:22] --- (14 headers 0 lines) ---
[2011-11-03 06:46:22] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4142' is now Reachable. (55ms / 2000ms)
[2011-11-03 06:46:22] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45414ebd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a2738ce
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 700f69796366f16d3717ece77a7687ba@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4113 (queued)
[2011-11-03 06:46:22] set_destination: Parsing <sip:4125@172.30.0.168:5060> for address/port to send to
[2011-11-03 06:46:22] set_destination: set destination to 172.30.0.168:5060
[2011-11-03 06:46:22] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b61946d
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as40b3dd21
To: "4125" <sip:4125@customer.com>;tag=887CE4DB-CFBDA658
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: e04a23b8-af109f31-58354f76@172.30.0.168
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4125
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4108 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4135 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4116 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4138 (queued)
[2011-11-03 06:46:22]   == Extension Changed 4142[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:46:22] Really destroying SIP dialog '4e007a445f5cb571765a94c466bcb0cc@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:22] Retransmitting #5 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:22] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0aaee423
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7aac0f75
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e9efc9066d9e6fa5f310e0856f681f7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72c611ce
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50e27b2a
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0be0c0dc001e1ad1173a40f548c90d27@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:22] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5520b03f
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as4b9f0e07
To: "4147" <sip:4147@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.183
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b61946d
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as40b3dd21
To: "4125" <sip:4125@customer.com>;tag=887CE4DB-CFBDA658
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: e04a23b8-af109f31-58354f76@172.30.0.168
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission cf08829-f486811a-8885c823@172.30.0.183 for seqno 107 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6384ms with no response
[2011-11-03 06:46:23] Really destroying SIP dialog 'cf08829-f486811a-8885c823@172.30.0.183' Method: SUBSCRIBE
[2011-11-03 06:46:23] Really destroying SIP dialog 'bed2c844-3152a6d9-614d6b2e@172.30.0.173' Method: REGISTER
[2011-11-03 06:46:23] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK108555bd
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as40f08f92
To: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37db833a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as62eb1ab4
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4af1034b2be7d5a14ba394147e5ecd90@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:23] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b61946d
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as40b3dd21
To: "4125" <sip:4125@customer.com>;tag=887CE4DB-CFBDA658
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: e04a23b8-af109f31-58354f76@172.30.0.168
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f17b40d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19ecb86a
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7187f29e49ff0db822009aad0bf8bbc0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:23] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f5726eb
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3ee1b4b1
To: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK63b4083f
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as31013ea5
To: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f355269
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0086ce3d
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3b5f80e86a42c54234f99a834c170906@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:23] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b61946d
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as40b3dd21
To: "4125" <sip:4125@customer.com>;tag=887CE4DB-CFBDA658
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: e04a23b8-af109f31-58354f76@172.30.0.168
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:23] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07dc3ada
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as732f9cef
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a9672f568a0ac143c4cf2680f62f21d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:23] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
OPTIONS sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK79c65c7c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6dbb025a
To: <sip:4136@172.30.0.172:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3696ed9d26d714c279a03a8873d8d46f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:23]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK79c65c7c
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6dbb025a
To: "4136" <sip:4136@172.30.0.172:5060>;tag=4D46AF72-348B28CB
CSeq: 102 OPTIONS
Call-ID: 3696ed9d26d714c279a03a8873d8d46f@172.30.0.10:5060
Contact: <sip:4136@172.30.0.172:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:23] --- (14 headers 0 lines) ---
[2011-11-03 06:46:23] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4136' is now Reachable. (44ms / 2000ms)
[2011-11-03 06:46:23] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533a3de2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as611f31d0
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6c2e6a7f3781f85b17b1c5a05d449c22@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:23] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45414ebd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a2738ce
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 700f69796366f16d3717ece77a7687ba@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4137 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4128 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:46:23]   == Extension Changed 4136[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:46:23] Really destroying SIP dialog '3696ed9d26d714c279a03a8873d8d46f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:24] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0aaee423
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7aac0f75
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e9efc9066d9e6fa5f310e0856f681f7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72c611ce
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50e27b2a
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0be0c0dc001e1ad1173a40f548c90d27@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e175806
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5fc896fe
To: "Office 106" <sip:4120@customer.com>;tag=3B9E553-C1793082
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
vent: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6398ms with no response
[2011-11-03 06:46:24] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37db833a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as62eb1ab4
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4af1034b2be7d5a14ba394147e5ecd90@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c190d43
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c883b31
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 442e056909ecbefb6d7cb827191bac25@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f17b40d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19ecb86a
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7187f29e49ff0db822009aad0bf8bbc0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK722dc804
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2b09121f
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 39aa30ca197035037facac9865778614@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5ded4d
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b61946d
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as40b3dd21
To: "4125" <sip:4125@customer.com>;tag=887CE4DB-CFBDA658
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: e04a23b8-af109f31-58354f76@172.30.0.168
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] Retransmitting #6 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK105fabbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2c27a83d
To: "4123" <sip:4123@customer.com>;tag=3E636E08-D924771F
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ea8b0c-8a3eb943-f9c5aff2@172.30.0.184
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ea8b0c-8a3eb943-f9c5aff2@172.30.0.184 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6392ms with no response
[2011-11-03 06:46:24] Retransmitting #9 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff6cbdd
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as284d87e4
To: "4118" <sip:4118@customer.com>;tag=E9AFD17D-87EBC8D2
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 28129ff1-31250646-5c60a25b@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f355269
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0086ce3d
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3b5f80e86a42c54234f99a834c170906@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07dc3ada
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as732f9cef
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a9672f568a0ac143c4cf2680f62f21d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25b4fe69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as61bf63b4
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 018ab9c561e273a7398a4ff35551c742@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK770c14be
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0788b28b
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5dff6d1347785262557ff36e0eac414a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533a3de2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as611f31d0
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6c2e6a7f3781f85b17b1c5a05d449c22@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK285dcb66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1165a920
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5d2b8236ae9913597001915c199785@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:24] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45414ebd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a2738ce
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 700f69796366f16d3717ece77a7687ba@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Really destroying SIP dialog 'c36bdc75-77e4d686-91aca9af@172.30.0.169' Method: REGISTER
[2011-11-03 06:46:24] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1765e245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2152db62
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0c84ef626674ad2e372a04221dc23473@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3edfa363
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0dd5de92
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ae673011c93bbfb58ee25067ea9e21f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:24] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0aaee423
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7aac0f75
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e9efc9066d9e6fa5f310e0856f681f7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72c611ce
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50e27b2a
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0be0c0dc001e1ad1173a40f548c90d27@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #5 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f5726eb
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3ee1b4b1
To: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:25] Retransmitting #5 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK63b4083f
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as31013ea5
To: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:25] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37db833a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as62eb1ab4
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4af1034b2be7d5a14ba394147e5ecd90@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c190d43
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c883b31
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 442e056909ecbefb6d7cb827191bac25@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f17b40d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19ecb86a
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7187f29e49ff0db822009aad0bf8bbc0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:25] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK722dc804
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2b09121f
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 39aa30ca197035037facac9865778614@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:25] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f355269
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0086ce3d
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3b5f80e86a42c54234f99a834c170906@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07dc3ada
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as732f9cef
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a9672f568a0ac143c4cf2680f62f21d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25b4fe69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as61bf63b4
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 018ab9c561e273a7398a4ff35551c742@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK770c14be
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0788b28b
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5dff6d1347785262557ff36e0eac414a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533a3de2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as611f31d0
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6c2e6a7f3781f85b17b1c5a05d449c22@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45414ebd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a2738ce
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 700f69796366f16d3717ece77a7687ba@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Really destroying SIP dialog '700f69796366f16d3717ece77a7687ba@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:25] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK285dcb66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1165a920
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5d2b8236ae9913597001915c199785@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1765e245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2152db62
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0c84ef626674ad2e372a04221dc23473@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0aaee423
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7aac0f75
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e9efc9066d9e6fa5f310e0856f681f7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:25] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4134' is now UNREACHABLE!  Last qualify: 62
[2011-11-03 06:46:25] Really destroying SIP dialog '3e9efc9066d9e6fa5f310e0856f681f7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:25] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72c611ce
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as50e27b2a
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0be0c0dc001e1ad1173a40f548c90d27@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:26] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:46:26] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3561bc02
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as44a9b59d
To: "4124" <sip:4124@customer.com>;tag=24F90BA9-B1ACFE6C
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4124
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4128 (queued)
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:46:26] set_destination: Parsing <sip:4168@172.30.0.187:5060> for address/port to send to
[2011-11-03 06:46:26] set_destination: set destination to 172.30.0.187:5060
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0501cd26
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as5fe9efcc
To: "4168" <sip:4168@customer.com>;tag=D0F8464B-5FA691DE
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4168
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:46:26] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:46:26] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23582062
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as45e86306
To: "Office 106" <sip:4120@customer.com>;tag=B5C80B61-8E43FC80
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4a5ea429-2efcc588-8586ea9f@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4120
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:46:26] set_destination: Parsing <sip:4149@172.30.0.188:5060> for address/port to send to
[2011-11-03 06:46:26] set_destination: set destination to 172.30.0.188:5060
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12d0a01b
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as766825a4
To: "4149" <sip:4149@customer.com>;tag=9D9811E7-1E6B7B2E
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 3ac02d8a-41ebf1d9-32249850@172.30.0.188
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4149
[2011-11-03 06:46:26] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:46:26] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d9a8323
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as3989b8ea
To: "4116" <sip:4116@customer.com>;tag=A6FF4B6F-C7B28FF0
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4ed404fb-2ca5289c-cd7342d5@172.30.0.199
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4116
[2011-11-03 06:46:26] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:26] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08e649ce
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26]   == Extension Changed 4134[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:26] Really destroying SIP dialog '0be0c0dc001e1ad1173a40f548c90d27@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:26] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3edfa363
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0dd5de92
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ae673011c93bbfb58ee25067ea9e21f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #5 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b61946d
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as40b3dd21
To: "4125" <sip:4125@customer.com>;tag=887CE4DB-CFBDA658
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: e04a23b8-af109f31-58354f76@172.30.0.168
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08e649ce
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
CSeq: 105 NOTIFY
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:26] --- (11 headers 0 lines) ---
[2011-11-03 06:46:26] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:26] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3561bc02
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as44a9b59d
To: "4124" <sip:4124@customer.com>;tag=24F90BA9-B1ACFE6C
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0501cd26
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as5fe9efcc
To: "4168" <sip:4168@customer.com>;tag=D0F8464B-5FA691DE
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23582062
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as45e86306
To: "Office 106" <sip:4120@customer.com>;tag=B5C80B61-8E43FC80
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4a5ea429-2efcc588-8586ea9f@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12d0a01b
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as766825a4
To: "4149" <sip:4149@customer.com>;tag=9D9811E7-1E6B7B2E
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 3ac02d8a-41ebf1d9-32249850@172.30.0.188
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d9a8323
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as3989b8ea
To: "4116" <sip:4116@customer.com>;tag=A6FF4B6F-C7B28FF0
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4ed404fb-2ca5289c-cd7342d5@172.30.0.199
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37db833a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as62eb1ab4
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4af1034b2be7d5a14ba394147e5ecd90@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Really destroying SIP dialog '4af1034b2be7d5a14ba394147e5ecd90@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c190d43
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c883b31
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 442e056909ecbefb6d7cb827191bac25@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK108555bd
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as40f08f92
To: "Leon Zhong" <sip:4134@customer.com>;tag=83F42F1D-8EE06BF2
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: f385b0a3-50915880-1318c659@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f17b40d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19ecb86a
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7187f29e49ff0db822009aad0bf8bbc0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3561bc02
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as44a9b59d
To: "4124" <sip:4124@customer.com>;tag=24F90BA9-B1ACFE6C
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0501cd26
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as5fe9efcc
To: "4168" <sip:4168@customer.com>;tag=D0F8464B-5FA691DE
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23582062
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as45e86306
To: "Office 106" <sip:4120@customer.com>;tag=B5C80B61-8E43FC80
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4a5ea429-2efcc588-8586ea9f@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12d0a01b
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as766825a4
To: "4149" <sip:4149@customer.com>;tag=9D9811E7-1E6B7B2E
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 3ac02d8a-41ebf1d9-32249850@172.30.0.188
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d9a8323
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as3989b8ea
To: "4116" <sip:4116@customer.com>;tag=A6FF4B6F-C7B28FF0
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4ed404fb-2ca5289c-cd7342d5@172.30.0.199
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK722dc804
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2b09121f
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 39aa30ca197035037facac9865778614@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #5 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission f385b0a3-50915880-1318c659@172.30.0.179 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6461ms with no response
[2011-11-03 06:46:26] Retransmitting #12 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5724f268
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7cb3e461
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 141d53ff4a0eb7d807514d503ba5b743@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f355269
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0086ce3d
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3b5f80e86a42c54234f99a834c170906@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Really destroying SIP dialog '3b5f80e86a42c54234f99a834c170906@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:26] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07dc3ada
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as732f9cef
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a9672f568a0ac143c4cf2680f62f21d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3561bc02
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as44a9b59d
To: "4124" <sip:4124@customer.com>;tag=24F90BA9-B1ACFE6C
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25b4fe69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as61bf63b4
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 018ab9c561e273a7398a4ff35551c742@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0501cd26
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as5fe9efcc
To: "4168" <sip:4168@customer.com>;tag=D0F8464B-5FA691DE
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23582062
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as45e86306
To: "Office 106" <sip:4120@customer.com>;tag=B5C80B61-8E43FC80
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4a5ea429-2efcc588-8586ea9f@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12d0a01b
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as766825a4
To: "4149" <sip:4149@customer.com>;tag=9D9811E7-1E6B7B2E
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 3ac02d8a-41ebf1d9-32249850@172.30.0.188
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d9a8323
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as3989b8ea
To: "4116" <sip:4116@customer.com>;tag=A6FF4B6F-C7B28FF0
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4ed404fb-2ca5289c-cd7342d5@172.30.0.199
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK770c14be
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0788b28b
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5dff6d1347785262557ff36e0eac414a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK743d34b9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71312b80
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 31086903309633c01f78c2f748ae0e1f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533a3de2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as611f31d0
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6c2e6a7f3781f85b17b1c5a05d449c22@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Really destroying SIP dialog '6c2e6a7f3781f85b17b1c5a05d449c22@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:26] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f1f628b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as616870a9
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f159521777b8ded2528e8c312611cc7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK285dcb66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1165a920
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5d2b8236ae9913597001915c199785@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #6 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1765e245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2152db62
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0c84ef626674ad2e372a04221dc23473@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:26] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3edfa363
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0dd5de92
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ae673011c93bbfb58ee25067ea9e21f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #5 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:27] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c190d43
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c883b31
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 442e056909ecbefb6d7cb827191bac25@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75fc5119
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25999b0d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65aab1f9362ba64b2b4df2011fdf627e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f17b40d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19ecb86a
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7187f29e49ff0db822009aad0bf8bbc0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Really destroying SIP dialog '7187f29e49ff0db822009aad0bf8bbc0@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:27] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK722dc804
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2b09121f
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 39aa30ca197035037facac9865778614@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Really destroying SIP dialog '191ae026-424cadc1-eaf44cdc@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:46:27] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d1f05dc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as165e8faf
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5f3c25b26c36a14401f657587d23e5f8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3561bc02
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as44a9b59d
To: "4124" <sip:4124@customer.com>;tag=24F90BA9-B1ACFE6C
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:27] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0501cd26
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as5fe9efcc
To: "4168" <sip:4168@customer.com>;tag=D0F8464B-5FA691DE
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:27] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23582062
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as45e86306
To: "Office 106" <sip:4120@customer.com>;tag=B5C80B61-8E43FC80
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4a5ea429-2efcc588-8586ea9f@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:27] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12d0a01b
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as766825a4
To: "4149" <sip:4149@customer.com>;tag=9D9811E7-1E6B7B2E
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 3ac02d8a-41ebf1d9-32249850@172.30.0.188
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:27] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d9a8323
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as3989b8ea
To: "4116" <sip:4116@customer.com>;tag=A6FF4B6F-C7B28FF0
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4ed404fb-2ca5289c-cd7342d5@172.30.0.199
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:27] Retransmitting #6 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f5ded4d
Max-Forwards: 70
From: <sip:4141@customer.com>;tag=as45e9cdb6
To: "Office 107" <sip:4141@customer.com>;tag=4325289E-C2FC860B
Contact: <sip:4141@172.30.0.10:5060>
Call-ID: 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4141">
<address uri="sip:4141@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:27] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5724f268
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7cb3e461
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 141d53ff4a0eb7d807514d503ba5b743@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181 for seqno 107 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6377ms with no response
[2011-11-03 06:46:27] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07dc3ada
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as732f9cef
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a9672f568a0ac143c4cf2680f62f21d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Really destroying SIP dialog '3a9672f568a0ac143c4cf2680f62f21d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:27] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25b4fe69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as61bf63b4
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 018ab9c561e273a7398a4ff35551c742@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK770c14be
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0788b28b
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5dff6d1347785262557ff36e0eac414a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK285dcb66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1165a920
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5d2b8236ae9913597001915c199785@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK743d34b9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71312b80
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 31086903309633c01f78c2f748ae0e1f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f1f628b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as616870a9
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f159521777b8ded2528e8c312611cc7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:27] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1765e245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2152db62
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0c84ef626674ad2e372a04221dc23473@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3edfa363
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0dd5de92
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ae673011c93bbfb58ee25067ea9e21f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c190d43
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c883b31
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 442e056909ecbefb6d7cb827191bac25@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4173' is now UNREACHABLE!  Last qualify: 42
[2011-11-03 06:46:28] Really destroying SIP dialog '442e056909ecbefb6d7cb827191bac25@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:46:28] set_destination: Parsing <sip:4168@172.30.0.187:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.187:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9cbf30
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as14980033
To: "4168" <sip:4168@customer.com>;tag=52960336-E860FCC1
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 8db2dcba-305421e5-dfa84f08@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4168
[2011-11-03 06:46:28] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK290aaeaa
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as5d0b152d
To: "4124" <sip:4124@customer.com>;tag=5B1D257F-6BBB7432
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b6cf293c-ba839a37-8ec697aa@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4124
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:28] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c0ecdf1
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as44d12cb7
To: "4135" <sip:4135@customer.com>;tag=2AB3F56B-CF74FA8
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: ef04609-b6df436e-e0c59a2f@172.30.0.177
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4135
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:28] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK629fb4a2
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as560d813a
To: "4147" <sip:4147@customer.com>;tag=2570BD84-F7F7D97D
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: d9cdeec-a9be8a5-5054ef36@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4147
[2011-11-03 06:46:28] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a0d7e09
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as0d315950
To: "4173" <sip:4173@customer.com>;tag=AC50AAD8-544DFE6F
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b8873c3a-9fa41161-e9fbb280@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4173
[2011-11-03 06:46:28] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b9a9054
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as1acba730
To: "4148" <sip:4148@customer.com>;tag=3175BF81-329839A0
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: c45d2e1b-2520298a-c4126d31@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4148
[2011-11-03 06:46:28] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1cdfe237
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as2176e8cd
To: "4136" <sip:4136@customer.com>;tag=855FBBE9-7000B282
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: da0ed330-32d92309-5c4af1a2@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4173[default-local] new state Unavailable for Notify User 4136
[2011-11-03 06:46:28] Retransmitting #6 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f5726eb
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3ee1b4b1
To: "4168" <sip:4168@customer.com>;tag=6540CCEE-C0E47D39
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1cdfe237
From: <sip:4173@customer.com>;tag=as2176e8cd
To: "4136" <sip:4136@customer.com>;tag=855FBBE9-7000B282
CSeq: 105 NOTIFY
Call-ID: da0ed330-32d92309-5c4af1a2@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:46:28] --- (11 headers 0 lines) ---
[2011-11-03 06:46:28] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:46:28] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 36f849fc-1ff2a8f7-2bd5446a@172.30.0.187 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6444ms with no response
[2011-11-03 06:46:28] Really destroying SIP dialog '36f849fc-1ff2a8f7-2bd5446a@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75fc5119
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25999b0d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65aab1f9362ba64b2b4df2011fdf627e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Retransmitting #6 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK63b4083f
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as31013ea5
To: "Office 113" <sip:4171@customer.com>;tag=3EEC643F-A78B1106
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: b8b4149-dd256cd8-6865693@172.30.0.165
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9cbf30
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as14980033
To: "4168" <sip:4168@customer.com>;tag=52960336-E860FCC1
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 8db2dcba-305421e5-dfa84f08@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK290aaeaa
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as5d0b152d
To: "4124" <sip:4124@customer.com>;tag=5B1D257F-6BBB7432
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b6cf293c-ba839a37-8ec697aa@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c0ecdf1
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as44d12cb7
To: "4135" <sip:4135@customer.com>;tag=2AB3F56B-CF74FA8
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: ef04609-b6df436e-e0c59a2f@172.30.0.177
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK629fb4a2
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as560d813a
To: "4147" <sip:4147@customer.com>;tag=2570BD84-F7F7D97D
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: d9cdeec-a9be8a5-5054ef36@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a0d7e09
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as0d315950
To: "4173" <sip:4173@customer.com>;tag=AC50AAD8-544DFE6F
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b8873c3a-9fa41161-e9fbb280@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK722dc804
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2b09121f
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 39aa30ca197035037facac9865778614@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4145' is now UNREACHABLE!  Last qualify: 74
[2011-11-03 06:46:28] Really destroying SIP dialog '39aa30ca197035037facac9865778614@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:28] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b8b4149-dd256cd8-6865693@172.30.0.165 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6407ms with no response
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b9a9054
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as1acba730
To: "4148" <sip:4148@customer.com>;tag=3175BF81-329839A0
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: c45d2e1b-2520298a-c4126d31@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:46:28] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b828d15
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as679e109b
To: "4124" <sip:4124@customer.com>;tag=EFFF03-65549956
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4124
[2011-11-03 06:46:28] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47df2854
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3a72c8db
To: "4115" <sip:4115@customer.com>;tag=6A840CA4-7E43813F
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 51ff2fc8-281ae663-daccee7e@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4115
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:46:28] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47511144
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as4d653910
To: "4147" <sip:4147@customer.com>;tag=E7EBDD1B-E23B33BC
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: ed5306d6-fa8a397f-dcc3c780@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4147
[2011-11-03 06:46:28] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d1f05dc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as165e8faf
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5f3c25b26c36a14401f657587d23e5f8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0df18116
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3275fa35
To: "Leon Zhong" <sip:4134@customer.com>;tag=8548EC13-2BA70B30
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: aeb59ec2-9f33d7f3-77070190@172.30.0.179
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4145[default-local] new state Unavailable for Notify User 4134
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9cbf30
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as14980033
To: "4168" <sip:4168@customer.com>;tag=52960336-E860FCC1
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 8db2dcba-305421e5-dfa84f08@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #10 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ff6cbdd
Max-Forwards: 70
From: <sip:4124@customer.com>;tag=as284d87e4
To: "4118" <sip:4118@customer.com>;tag=E9AFD17D-87EBC8D2
Contact: <sip:4124@172.30.0.10:5060>
Call-ID: 28129ff1-31250646-5c60a25b@172.30.0.173
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4124">
<address uri="sip:4124@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK290aaeaa
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as5d0b152d
To: "4124" <sip:4124@customer.com>;tag=5B1D257F-6BBB7432
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b6cf293c-ba839a37-8ec697aa@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c0ecdf1
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as44d12cb7
To: "4135" <sip:4135@customer.com>;tag=2AB3F56B-CF74FA8
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: ef04609-b6df436e-e0c59a2f@172.30.0.177
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK629fb4a2
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as560d813a
To: "4147" <sip:4147@customer.com>;tag=2570BD84-F7F7D97D
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: d9cdeec-a9be8a5-5054ef36@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a0d7e09
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as0d315950
To: "4173" <sip:4173@customer.com>;tag=AC50AAD8-544DFE6F
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b8873c3a-9fa41161-e9fbb280@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b828d15
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as679e109b
To: "4124" <sip:4124@customer.com>;tag=EFFF03-65549956
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b9a9054
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as1acba730
To: "4148" <sip:4148@customer.com>;tag=3175BF81-329839A0
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: c45d2e1b-2520298a-c4126d31@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47df2854
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3a72c8db
To: "4115" <sip:4115@customer.com>;tag=6A840CA4-7E43813F
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 51ff2fc8-281ae663-daccee7e@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47511144
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as4d653910
To: "4147" <sip:4147@customer.com>;tag=E7EBDD1B-E23B33BC
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: ed5306d6-fa8a397f-dcc3c780@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5724f268
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7cb3e461
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 141d53ff4a0eb7d807514d503ba5b743@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0df18116
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3275fa35
To: "Leon Zhong" <sip:4134@customer.com>;tag=8548EC13-2BA70B30
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: aeb59ec2-9f33d7f3-77070190@172.30.0.179
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25b4fe69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as61bf63b4
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 018ab9c561e273a7398a4ff35551c742@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Really destroying SIP dialog '018ab9c561e273a7398a4ff35551c742@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b828d15
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as679e109b
To: "4124" <sip:4124@customer.com>;tag=EFFF03-65549956
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47df2854
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3a72c8db
To: "4115" <sip:4115@customer.com>;tag=6A840CA4-7E43813F
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 51ff2fc8-281ae663-daccee7e@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47511144
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as4d653910
To: "4147" <sip:4147@customer.com>;tag=E7EBDD1B-E23B33BC
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: ed5306d6-fa8a397f-dcc3c780@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK770c14be
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0788b28b
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5dff6d1347785262557ff36e0eac414a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4149' is now UNREACHABLE!  Last qualify: 56
[2011-11-03 06:46:28] Really destroying SIP dialog '5dff6d1347785262557ff36e0eac414a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:28] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:46:28] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:46:28] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25023910
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as74786235
To: "Office 131" <sip:4113@customer.com>;tag=7C2FCF91-B98A75D8
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: cf747800-84c73897-29c17efe@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28]   == Extension Changed 4149[default-local] new state Unavailable for Notify User 4113
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0df18116
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3275fa35
To: "Leon Zhong" <sip:4134@customer.com>;tag=8548EC13-2BA70B30
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: aeb59ec2-9f33d7f3-77070190@172.30.0.179
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #7 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:28] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK285dcb66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1165a920
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5d2b8236ae9913597001915c199785@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Really destroying SIP dialog '1d5d2b8236ae9913597001915c199785@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK743d34b9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71312b80
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 31086903309633c01f78c2f748ae0e1f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f1f628b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as616870a9
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f159521777b8ded2528e8c312611cc7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1765e245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2152db62
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0c84ef626674ad2e372a04221dc23473@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:28] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4137' is now UNREACHABLE!  Last qualify: 44
[2011-11-03 06:46:28] Really destroying SIP dialog '0c84ef626674ad2e372a04221dc23473@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9cbf30
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as14980033
To: "4168" <sip:4168@customer.com>;tag=52960336-E860FCC1
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 8db2dcba-305421e5-dfa84f08@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4128 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4116 (queued)
[2011-11-03 06:46:29]   == Extension Changed 4137[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK290aaeaa
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as5d0b152d
To: "4124" <sip:4124@customer.com>;tag=5B1D257F-6BBB7432
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b6cf293c-ba839a37-8ec697aa@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c0ecdf1
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as44d12cb7
To: "4135" <sip:4135@customer.com>;tag=2AB3F56B-CF74FA8
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: ef04609-b6df436e-e0c59a2f@172.30.0.177
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK629fb4a2
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as560d813a
To: "4147" <sip:4147@customer.com>;tag=2570BD84-F7F7D97D
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: d9cdeec-a9be8a5-5054ef36@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a0d7e09
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as0d315950
To: "4173" <sip:4173@customer.com>;tag=AC50AAD8-544DFE6F
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b8873c3a-9fa41161-e9fbb280@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25023910
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as74786235
To: "Office 131" <sip:4113@customer.com>;tag=7C2FCF91-B98A75D8
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: cf747800-84c73897-29c17efe@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3edfa363
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0dd5de92
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ae673011c93bbfb58ee25067ea9e21f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:29] Really destroying SIP dialog '2ae673011c93bbfb58ee25067ea9e21f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b9a9054
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as1acba730
To: "4148" <sip:4148@customer.com>;tag=3175BF81-329839A0
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: c45d2e1b-2520298a-c4126d31@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 28129ff1-31250646-5c60a25b@172.30.0.173 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32018ms with no response
[2011-11-03 06:46:29] Really destroying SIP dialog '28129ff1-31250646-5c60a25b@172.30.0.173' Method: SUBSCRIBE
[2011-11-03 06:46:29] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3561bc02
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as44a9b59d
To: "4124" <sip:4124@customer.com>;tag=24F90BA9-B1ACFE6C
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #5 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0501cd26
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as5fe9efcc
To: "4168" <sip:4168@customer.com>;tag=D0F8464B-5FA691DE
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23582062
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as45e86306
To: "Office 106" <sip:4120@customer.com>;tag=B5C80B61-8E43FC80
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4a5ea429-2efcc588-8586ea9f@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #5 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12d0a01b
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as766825a4
To: "4149" <sip:4149@customer.com>;tag=9D9811E7-1E6B7B2E
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 3ac02d8a-41ebf1d9-32249850@172.30.0.188
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #5 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d9a8323
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as3989b8ea
To: "4116" <sip:4116@customer.com>;tag=A6FF4B6F-C7B28FF0
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4ed404fb-2ca5289c-cd7342d5@172.30.0.199
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25023910
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as74786235
To: "Office 131" <sip:4113@customer.com>;tag=7C2FCF91-B98A75D8
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: cf747800-84c73897-29c17efe@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b828d15
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as679e109b
To: "4124" <sip:4124@customer.com>;tag=EFFF03-65549956
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47df2854
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3a72c8db
To: "4115" <sip:4115@customer.com>;tag=6A840CA4-7E43813F
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 51ff2fc8-281ae663-daccee7e@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47511144
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as4d653910
To: "4147" <sip:4147@customer.com>;tag=E7EBDD1B-E23B33BC
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: ed5306d6-fa8a397f-dcc3c780@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0df18116
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3275fa35
To: "Leon Zhong" <sip:4134@customer.com>;tag=8548EC13-2BA70B30
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: aeb59ec2-9f33d7f3-77070190@172.30.0.179
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #6 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b61946d
Max-Forwards: 70
From: <sip:4142@customer.com>;tag=as40b3dd21
To: "4125" <sip:4125@customer.com>;tag=887CE4DB-CFBDA658
Contact: <sip:4142@172.30.0.10:5060>
Call-ID: e04a23b8-af109f31-58354f76@172.30.0.168
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4142">
<address uri="sip:4142@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e04a23b8-af109f31-58354f76@172.30.0.168 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6377ms with no response
[2011-11-03 06:46:29] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75fc5119
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25999b0d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65aab1f9362ba64b2b4df2011fdf627e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:29] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d1f05dc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as165e8faf
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5f3c25b26c36a14401f657587d23e5f8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25023910
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as74786235
To: "Office 131" <sip:4113@customer.com>;tag=7C2FCF91-B98A75D8
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: cf747800-84c73897-29c17efe@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5724f268
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7cb3e461
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 141d53ff4a0eb7d807514d503ba5b743@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9cbf30
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as14980033
To: "4168" <sip:4168@customer.com>;tag=52960336-E860FCC1
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 8db2dcba-305421e5-dfa84f08@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK290aaeaa
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as5d0b152d
To: "4124" <sip:4124@customer.com>;tag=5B1D257F-6BBB7432
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b6cf293c-ba839a37-8ec697aa@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c0ecdf1
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as44d12cb7
To: "4135" <sip:4135@customer.com>;tag=2AB3F56B-CF74FA8
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: ef04609-b6df436e-e0c59a2f@172.30.0.177
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK629fb4a2
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as560d813a
To: "4147" <sip:4147@customer.com>;tag=2570BD84-F7F7D97D
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: d9cdeec-a9be8a5-5054ef36@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a0d7e09
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as0d315950
To: "4173" <sip:4173@customer.com>;tag=AC50AAD8-544DFE6F
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b8873c3a-9fa41161-e9fbb280@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b9a9054
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as1acba730
To: "4148" <sip:4148@customer.com>;tag=3175BF81-329839A0
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: c45d2e1b-2520298a-c4126d31@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK743d34b9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71312b80
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 31086903309633c01f78c2f748ae0e1f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:29] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f1f628b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as616870a9
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f159521777b8ded2528e8c312611cc7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:29] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b828d15
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as679e109b
To: "4124" <sip:4124@customer.com>;tag=EFFF03-65549956
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47df2854
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3a72c8db
To: "4115" <sip:4115@customer.com>;tag=6A840CA4-7E43813F
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 51ff2fc8-281ae663-daccee7e@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47511144
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as4d653910
To: "4147" <sip:4147@customer.com>;tag=E7EBDD1B-E23B33BC
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: ed5306d6-fa8a397f-dcc3c780@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0df18116
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3275fa35
To: "Leon Zhong" <sip:4134@customer.com>;tag=8548EC13-2BA70B30
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: aeb59ec2-9f33d7f3-77070190@172.30.0.179
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Really destroying SIP dialog 'd4107dea-3da43fa1-2d77fb94@172.30.0.190' Method: REGISTER
[2011-11-03 06:46:30] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75fc5119
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25999b0d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65aab1f9362ba64b2b4df2011fdf627e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:30] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25023910
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as74786235
To: "Office 131" <sip:4113@customer.com>;tag=7C2FCF91-B98A75D8
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: cf747800-84c73897-29c17efe@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Retransmitting #6 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Really destroying SIP dialog '3c6ccdbd-7b4d0cbc-9c140473@172.30.0.180' Method: SUBSCRIBE
[2011-11-03 06:46:30] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d1f05dc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as165e8faf
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5f3c25b26c36a14401f657587d23e5f8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:30] Retransmitting #13 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581d5e43
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as3a04742c
To: "4123" <sip:4123@customer.com>;tag=73C9538A-41256731
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 78afa8a4-3476919b-bcfa090a@172.30.0.184
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5724f268
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7cb3e461
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 141d53ff4a0eb7d807514d503ba5b743@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:30] Really destroying SIP dialog '141d53ff4a0eb7d807514d503ba5b743@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:30] Really destroying SIP dialog '8c353181-8bf69970-4ed0d70b@172.30.0.165' Method: REGISTER
[2011-11-03 06:46:30] Really destroying SIP dialog 'ea8b0c-8a3eb943-f9c5aff2@172.30.0.184' Method: SUBSCRIBE
[2011-11-03 06:46:30] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK743d34b9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71312b80
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 31086903309633c01f78c2f748ae0e1f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:30] Really destroying SIP dialog '31086903309633c01f78c2f748ae0e1f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:30] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f1f628b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as616870a9
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f159521777b8ded2528e8c312611cc7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:30] Really destroying SIP dialog '4f159521777b8ded2528e8c312611cc7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:30] Retransmitting #7 (no NAT) to 172.30.0.182:5060:
NOTIFY sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0cbdd78a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as51112dce
To: "Office 108" <sip:4128@customer.com>;tag=D61E4A49-A9AB663A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: cf08829-f486811a-8885c823@172.30.0.182
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4128@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:30] Retransmitting #6 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75fc5119
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25999b0d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65aab1f9362ba64b2b4df2011fdf627e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:31] Really destroying SIP dialog '65aab1f9362ba64b2b4df2011fdf627e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9cbf30
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as14980033
To: "4168" <sip:4168@customer.com>;tag=52960336-E860FCC1
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 8db2dcba-305421e5-dfa84f08@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK290aaeaa
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as5d0b152d
To: "4124" <sip:4124@customer.com>;tag=5B1D257F-6BBB7432
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b6cf293c-ba839a37-8ec697aa@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c0ecdf1
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as44d12cb7
To: "4135" <sip:4135@customer.com>;tag=2AB3F56B-CF74FA8
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: ef04609-b6df436e-e0c59a2f@172.30.0.177
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK629fb4a2
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as560d813a
To: "4147" <sip:4147@customer.com>;tag=2570BD84-F7F7D97D
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: d9cdeec-a9be8a5-5054ef36@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a0d7e09
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as0d315950
To: "4173" <sip:4173@customer.com>;tag=AC50AAD8-544DFE6F
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b8873c3a-9fa41161-e9fbb280@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b9a9054
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as1acba730
To: "4148" <sip:4148@customer.com>;tag=3175BF81-329839A0
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: c45d2e1b-2520298a-c4126d31@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d1f05dc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as165e8faf
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5f3c25b26c36a14401f657587d23e5f8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:31] Really destroying SIP dialog '5f3c25b26c36a14401f657587d23e5f8@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b828d15
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as679e109b
To: "4124" <sip:4124@customer.com>;tag=EFFF03-65549956
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47df2854
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3a72c8db
To: "4115" <sip:4115@customer.com>;tag=6A840CA4-7E43813F
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 51ff2fc8-281ae663-daccee7e@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47511144
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as4d653910
To: "4147" <sip:4147@customer.com>;tag=E7EBDD1B-E23B33BC
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: ed5306d6-fa8a397f-dcc3c780@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0df18116
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3275fa35
To: "Leon Zhong" <sip:4134@customer.com>;tag=8548EC13-2BA70B30
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: aeb59ec2-9f33d7f3-77070190@172.30.0.179
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:31] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission cf08829-f486811a-8885c823@172.30.0.182 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 16877ms with no response
[2011-11-03 06:46:31] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25023910
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as74786235
To: "Office 131" <sip:4113@customer.com>;tag=7C2FCF91-B98A75D8
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: cf747800-84c73897-29c17efe@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] Really destroying SIP dialog 'bed2c844-3152a6d9-614d6b2e@172.30.0.185' Method: REGISTER
[2011-11-03 06:46:32] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3561bc02
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as44a9b59d
To: "4124" <sip:4124@customer.com>;tag=24F90BA9-B1ACFE6C
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] Retransmitting #6 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0501cd26
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as5fe9efcc
To: "4168" <sip:4168@customer.com>;tag=D0F8464B-5FA691DE
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 441b21f5-596a0d98-ee70e5f3@172.30.0.187
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23582062
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as45e86306
To: "Office 106" <sip:4120@customer.com>;tag=B5C80B61-8E43FC80
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4a5ea429-2efcc588-8586ea9f@172.30.0.180
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] Retransmitting #6 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12d0a01b
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as766825a4
To: "4149" <sip:4149@customer.com>;tag=9D9811E7-1E6B7B2E
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 3ac02d8a-41ebf1d9-32249850@172.30.0.188
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] Retransmitting #6 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d9a8323
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as3989b8ea
To: "4116" <sip:4116@customer.com>;tag=A6FF4B6F-C7B28FF0
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 4ed404fb-2ca5289c-cd7342d5@172.30.0.199
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 441b21f5-596a0d98-ee70e5f3@172.30.0.187 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6379ms with no response
[2011-11-03 06:46:32] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 441b21f5-596a0d98-ee70e5f3@172.30.0.164 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6434ms with no response
[2011-11-03 06:46:32] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 4a5ea429-2efcc588-8586ea9f@172.30.0.180 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6416ms with no response
[2011-11-03 06:46:32] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3ac02d8a-41ebf1d9-32249850@172.30.0.188 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6408ms with no response
[2011-11-03 06:46:32] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 4ed404fb-2ca5289c-cd7342d5@172.30.0.199 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6404ms with no response
[2011-11-03 06:46:32] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] Really destroying SIP dialog 'f385b0a3-50915880-1318c659@172.30.0.179' Method: SUBSCRIBE
[2011-11-03 06:46:32] Retransmitting #8 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:32] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 78afa8a4-3476919b-bcfa090a@172.30.0.184 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 47551ms with no response
[2011-11-03 06:46:33] Really destroying SIP dialog '7cc87e10-62587e63-354d590a@172.30.0.191' Method: REGISTER
[2011-11-03 06:46:33] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:33] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Really destroying SIP dialog '12ffaa6a-a4ccd2c7-1b7d9ca8@172.30.0.181' Method: SUBSCRIBE
[2011-11-03 06:46:34] Retransmitting #7 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.187:5060:
NOTIFY sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e9cbf30
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as14980033
To: "4168" <sip:4168@customer.com>;tag=52960336-E860FCC1
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: 8db2dcba-305421e5-dfa84f08@172.30.0.187
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4168@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK290aaeaa
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as5d0b152d
To: "4124" <sip:4124@customer.com>;tag=5B1D257F-6BBB7432
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b6cf293c-ba839a37-8ec697aa@172.30.0.164
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1c0ecdf1
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as44d12cb7
To: "4135" <sip:4135@customer.com>;tag=2AB3F56B-CF74FA8
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: ef04609-b6df436e-e0c59a2f@172.30.0.177
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK629fb4a2
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as560d813a
To: "4147" <sip:4147@customer.com>;tag=2570BD84-F7F7D97D
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: d9cdeec-a9be8a5-5054ef36@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a0d7e09
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as0d315950
To: "4173" <sip:4173@customer.com>;tag=AC50AAD8-544DFE6F
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: b8873c3a-9fa41161-e9fbb280@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b9a9054
Max-Forwards: 70
From: <sip:4173@customer.com>;tag=as1acba730
To: "4148" <sip:4148@customer.com>;tag=3175BF81-329839A0
Contact: <sip:4173@172.30.0.10:5060>
Call-ID: c45d2e1b-2520298a-c4126d31@172.30.0.192
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4173">
<address uri="sip:4173@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b6cf293c-ba839a37-8ec697aa@172.30.0.164 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6385ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ef04609-b6df436e-e0c59a2f@172.30.0.177 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6380ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 8db2dcba-305421e5-dfa84f08@172.30.0.187 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6429ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission d9cdeec-a9be8a5-5054ef36@172.30.0.183 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c45d2e1b-2520298a-c4126d31@172.30.0.192 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6383ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b8873c3a-9fa41161-e9fbb280@172.30.0.178 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b828d15
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as679e109b
To: "4124" <sip:4124@customer.com>;tag=EFFF03-65549956
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: e9d3782e-2da61a79-58cc69bc@172.30.0.164
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Really destroying SIP dialog 'b8b4149-dd256cd8-6865693@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47df2854
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3a72c8db
To: "4115" <sip:4115@customer.com>;tag=6A840CA4-7E43813F
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: 51ff2fc8-281ae663-daccee7e@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47511144
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as4d653910
To: "4147" <sip:4147@customer.com>;tag=E7EBDD1B-E23B33BC
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: ed5306d6-fa8a397f-dcc3c780@172.30.0.183
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0df18116
Max-Forwards: 70
From: <sip:4145@customer.com>;tag=as3275fa35
To: "Leon Zhong" <sip:4134@customer.com>;tag=8548EC13-2BA70B30
Contact: <sip:4145@172.30.0.10:5060>
Call-ID: aeb59ec2-9f33d7f3-77070190@172.30.0.179
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4145">
<address uri="sip:4145@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 51ff2fc8-281ae663-daccee7e@172.30.0.197 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6393ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ed5306d6-fa8a397f-dcc3c780@172.30.0.183 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6387ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e9d3782e-2da61a79-58cc69bc@172.30.0.164 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6407ms with no response
[2011-11-03 06:46:34] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission aeb59ec2-9f33d7f3-77070190@172.30.0.179 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6379ms with no response
[2011-11-03 06:46:34] Retransmitting #7 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:35] Retransmitting #6 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25023910
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as74786235
To: "Office 131" <sip:4113@customer.com>;tag=7C2FCF91-B98A75D8
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: cf747800-84c73897-29c17efe@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:35] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission cf747800-84c73897-29c17efe@172.30.0.195 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2011-11-03 06:46:35] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22d20eff
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as769c66d1
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5e4f36076645309961b1b0e901cd31d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:35] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5eca8e87
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63391be3
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 148e2af02b08439d2ccdf84e7d7e6cec@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:36] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0528591c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1094234f
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1df9bcae379710565d05c5630cfcea12@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:36] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27157d33
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca63f92
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e20a09134cd9eba37e5e4313b1e45cb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:36] Retransmitting #7 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:36] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50f7f723
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as500b3b54
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 704679b340a2cfe86bb799322812eb6b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:36] Retransmitting #7 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:36] Retransmitting #9 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:36] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72b7496f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26bb4d1a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65b9a5e377d10ce54ee6267633ba53ab@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:36] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22d20eff
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as769c66d1
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5e4f36076645309961b1b0e901cd31d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:36] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5eca8e87
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63391be3
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 148e2af02b08439d2ccdf84e7d7e6cec@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0528591c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1094234f
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1df9bcae379710565d05c5630cfcea12@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27157d33
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca63f92
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e20a09134cd9eba37e5e4313b1e45cb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Retransmitting #7 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:37] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e03af6a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as13037416
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 37c113cc646a5d14615565856118bfa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Retransmitting #7 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:37] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50f7f723
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as500b3b54
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 704679b340a2cfe86bb799322812eb6b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cf7d457
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b4936e9
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2c43d8e3461eb3e3676c25313fae5aff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22d20eff
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as769c66d1
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5e4f36076645309961b1b0e901cd31d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72b7496f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26bb4d1a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65b9a5e377d10ce54ee6267633ba53ab@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:37] Really destroying SIP dialog 'b85bd9a5-c8e45da-e219d72b@172.30.0.177' Method: REGISTER
[2011-11-03 06:46:38] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5eca8e87
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63391be3
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 148e2af02b08439d2ccdf84e7d7e6cec@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0528591c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1094234f
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1df9bcae379710565d05c5630cfcea12@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27157d33
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca63f92
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e20a09134cd9eba37e5e4313b1e45cb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23bb9428
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3e03053f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7cdfeb85251235b324d2932f52d67843@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Retransmitting #8 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:38] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e03af6a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as13037416
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 37c113cc646a5d14615565856118bfa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50e7a1d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as021e54ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12b8cb8400a2850628c7c05623e31ef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50f7f723
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as500b3b54
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 704679b340a2cfe86bb799322812eb6b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cf7d457
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b4936e9
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2c43d8e3461eb3e3676c25313fae5aff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74648ec9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e195a68
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3284ad09ed18b92c533b236e61a04d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581060e7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2f6d7fdb
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2e7b8d42326c971c075c259f47618df9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22d20eff
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as769c66d1
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5e4f36076645309961b1b0e901cd31d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72b7496f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26bb4d1a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65b9a5e377d10ce54ee6267633ba53ab@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:38] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f4e48ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as46f66115
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f2fce971585525b321b0de72427dc26@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #8 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:39] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5eca8e87
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63391be3
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 148e2af02b08439d2ccdf84e7d7e6cec@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK022ea5e3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as628692f1
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 21afe6d5285a07bc00fb0ccd41883842@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aed1f67
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as01c57371
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c7c108c2ba617283b43ca1878d121a2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0528591c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1094234f
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1df9bcae379710565d05c5630cfcea12@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27157d33
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca63f92
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e20a09134cd9eba37e5e4313b1e45cb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23bb9428
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3e03053f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7cdfeb85251235b324d2932f52d67843@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e03af6a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as13037416
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 37c113cc646a5d14615565856118bfa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50e7a1d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as021e54ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12b8cb8400a2850628c7c05623e31ef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50f7f723
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as500b3b54
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 704679b340a2cfe86bb799322812eb6b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cf7d457
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b4936e9
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2c43d8e3461eb3e3676c25313fae5aff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74648ec9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e195a68
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3284ad09ed18b92c533b236e61a04d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581060e7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2f6d7fdb
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2e7b8d42326c971c075c259f47618df9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK22d20eff
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as769c66d1
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5e4f36076645309961b1b0e901cd31d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Really destroying SIP dialog '5e4f36076645309961b1b0e901cd31d9@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:39] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72b7496f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26bb4d1a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65b9a5e377d10ce54ee6267633ba53ab@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:39] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f4e48ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as46f66115
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f2fce971585525b321b0de72427dc26@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5eca8e87
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63391be3
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 148e2af02b08439d2ccdf84e7d7e6cec@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Really destroying SIP dialog '148e2af02b08439d2ccdf84e7d7e6cec@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:40] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK022ea5e3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as628692f1
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 21afe6d5285a07bc00fb0ccd41883842@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aed1f67
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as01c57371
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c7c108c2ba617283b43ca1878d121a2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0528591c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1094234f
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1df9bcae379710565d05c5630cfcea12@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Really destroying SIP dialog '1df9bcae379710565d05c5630cfcea12@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:40] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27157d33
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca63f92
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e20a09134cd9eba37e5e4313b1e45cb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Really destroying SIP dialog '3e20a09134cd9eba37e5e4313b1e45cb@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:40] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23bb9428
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3e03053f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7cdfeb85251235b324d2932f52d67843@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #8 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:40] Reliably Transmitting (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK278b3db5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as655b164f
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6270165b17e3420906d26b693be4abf1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e03af6a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as13037416
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 37c113cc646a5d14615565856118bfa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40]
<--- SIP read from UDP:172.30.0.186:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK278b3db5
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as655b164f
To: "4169" <sip:4169@172.30.0.186:5060>;tag=C8EBDF68-1CE6F507
CSeq: 102 OPTIONS
Call-ID: 6270165b17e3420906d26b693be4abf1@172.30.0.10:5060
Contact: <sip:4169@172.30.0.186:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:46:40] --- (14 headers 0 lines) ---
[2011-11-03 06:46:40] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50e7a1d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as021e54ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12b8cb8400a2850628c7c05623e31ef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Really destroying SIP dialog '6270165b17e3420906d26b693be4abf1@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:40] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4529df98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b46eaed
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 32d624202bac7c8f63c388ce38ff2cb6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:46:40] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:46:40] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:40] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50f7f723
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as500b3b54
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 704679b340a2cfe86bb799322812eb6b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Really destroying SIP dialog '704679b340a2cfe86bb799322812eb6b@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:40] Retransmitting #8 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:40] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cf7d457
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b4936e9
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2c43d8e3461eb3e3676c25313fae5aff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74648ec9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e195a68
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3284ad09ed18b92c533b236e61a04d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #10 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3acc95b4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as59b14bbd
To: "4118" <sip:4118@customer.com>;tag=2A462A81-C9C55CD6
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 250d68f8-3904e68d-145089e2@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:40] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581060e7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2f6d7fdb
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2e7b8d42326c971c075c259f47618df9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK72b7496f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26bb4d1a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 65b9a5e377d10ce54ee6267633ba53ab@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Really destroying SIP dialog '65b9a5e377d10ce54ee6267633ba53ab@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:40] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3970dec3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69183c3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 26776e4571f6cf1139d07a9f222858ca@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:40] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c95905f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19456f6d
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0d1483aa692be05a22fb7d0e7fc66f7c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f4e48ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as46f66115
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f2fce971585525b321b0de72427dc26@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK022ea5e3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as628692f1
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 21afe6d5285a07bc00fb0ccd41883842@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Really destroying SIP dialog '5a7cf879-53c09394-e0e2c02f@172.30.0.197' Method: REGISTER
[2011-11-03 06:46:41] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aed1f67
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as01c57371
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c7c108c2ba617283b43ca1878d121a2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:41] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23bb9428
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3e03053f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7cdfeb85251235b324d2932f52d67843@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 250d68f8-3904e68d-145089e2@172.30.0.173 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2011-11-03 06:46:41] Really destroying SIP dialog '250d68f8-3904e68d-145089e2@172.30.0.173' Method: SUBSCRIBE
[2011-11-03 06:46:41] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d43a7c9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as68c0160d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 546f0b9a3c8ccfeb4c9341345a688a63@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Retransmitting #8 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:41] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6e03af6a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as13037416
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 37c113cc646a5d14615565856118bfa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Really destroying SIP dialog '37c113cc646a5d14615565856118bfa4@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:41] Retransmitting #8 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:41] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50e7a1d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as021e54ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12b8cb8400a2850628c7c05623e31ef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK05bc8526
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as275f5957
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 270cd8c464bfc4531a3412181807b851@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4529df98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b46eaed
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 32d624202bac7c8f63c388ce38ff2cb6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cf7d457
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5b4936e9
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2c43d8e3461eb3e3676c25313fae5aff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Really destroying SIP dialog '2c43d8e3461eb3e3676c25313fae5aff@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:41] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74648ec9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e195a68
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3284ad09ed18b92c533b236e61a04d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:41] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581060e7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2f6d7fdb
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2e7b8d42326c971c075c259f47618df9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f4e48ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as46f66115
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f2fce971585525b321b0de72427dc26@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3970dec3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69183c3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 26776e4571f6cf1139d07a9f222858ca@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c95905f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19456f6d
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0d1483aa692be05a22fb7d0e7fc66f7c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK022ea5e3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as628692f1
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 21afe6d5285a07bc00fb0ccd41883842@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aed1f67
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as01c57371
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c7c108c2ba617283b43ca1878d121a2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:42] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23bb9428
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3e03053f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7cdfeb85251235b324d2932f52d67843@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Really destroying SIP dialog '7cdfeb85251235b324d2932f52d67843@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:42] Retransmitting #9 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:42] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d43a7c9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as68c0160d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 546f0b9a3c8ccfeb4c9341345a688a63@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50e7a1d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as021e54ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 12b8cb8400a2850628c7c05623e31ef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Really destroying SIP dialog '12b8cb8400a2850628c7c05623e31ef7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:42] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4529df98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b46eaed
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 32d624202bac7c8f63c388ce38ff2cb6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK05bc8526
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as275f5957
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 270cd8c464bfc4531a3412181807b851@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74648ec9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e195a68
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3284ad09ed18b92c533b236e61a04d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Really destroying SIP dialog '7a3284ad09ed18b92c533b236e61a04d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:42] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK581060e7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2f6d7fdb
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2e7b8d42326c971c075c259f47618df9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:42] Really destroying SIP dialog '2e7b8d42326c971c075c259f47618df9@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:42] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f4e48ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as46f66115
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f2fce971585525b321b0de72427dc26@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Really destroying SIP dialog '6f2fce971585525b321b0de72427dc26@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:43] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3970dec3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69183c3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 26776e4571f6cf1139d07a9f222858ca@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Retransmitting #9 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:43] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c95905f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19456f6d
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0d1483aa692be05a22fb7d0e7fc66f7c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK022ea5e3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as628692f1
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 21afe6d5285a07bc00fb0ccd41883842@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Really destroying SIP dialog '21afe6d5285a07bc00fb0ccd41883842@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:43] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aed1f67
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as01c57371
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c7c108c2ba617283b43ca1878d121a2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Really destroying SIP dialog '4c7c108c2ba617283b43ca1878d121a2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:43] Really destroying SIP dialog 'b039439d-79868a24-bf6a4b7b@172.30.0.195' Method: REGISTER
[2011-11-03 06:46:43] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d43a7c9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as68c0160d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 546f0b9a3c8ccfeb4c9341345a688a63@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4529df98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b46eaed
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 32d624202bac7c8f63c388ce38ff2cb6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK05bc8526
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as275f5957
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 270cd8c464bfc4531a3412181807b851@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:43] Really destroying SIP dialog 'c2f6c859-ff2f4a9c-155d1a97@172.30.0.164' Method: REGISTER
[2011-11-03 06:46:43] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3970dec3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69183c3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 26776e4571f6cf1139d07a9f222858ca@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:44] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c95905f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19456f6d
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0d1483aa692be05a22fb7d0e7fc66f7c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:44] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:44] Retransmitting #9 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:44] Really destroying SIP dialog 'b85bd9a5-c8e45da-e219d72b@172.30.0.168' Method: REGISTER
[2011-11-03 06:46:44] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d43a7c9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as68c0160d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 546f0b9a3c8ccfeb4c9341345a688a63@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:44] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4529df98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b46eaed
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 32d624202bac7c8f63c388ce38ff2cb6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:44] Really destroying SIP dialog '32d624202bac7c8f63c388ce38ff2cb6@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:44] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK05bc8526
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as275f5957
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 270cd8c464bfc4531a3412181807b851@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:44] Retransmitting #9 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:44] Really destroying SIP dialog 'c2f6c859-ff2f4a9c-155d1a97@172.30.0.170' Method: REGISTER
[2011-11-03 06:46:44] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3970dec3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69183c3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 26776e4571f6cf1139d07a9f222858ca@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:45] Really destroying SIP dialog '26776e4571f6cf1139d07a9f222858ca@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:45] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c95905f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as19456f6d
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0d1483aa692be05a22fb7d0e7fc66f7c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:45] Really destroying SIP dialog '0d1483aa692be05a22fb7d0e7fc66f7c@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:45] Retransmitting #9 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:45] Retransmitting #9 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:45] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d43a7c9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as68c0160d
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 546f0b9a3c8ccfeb4c9341345a688a63@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:45] Really destroying SIP dialog '546f0b9a3c8ccfeb4c9341345a688a63@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:45] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK05bc8526
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as275f5957
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 270cd8c464bfc4531a3412181807b851@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:45] Really destroying SIP dialog '270cd8c464bfc4531a3412181807b851@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:46] Retransmitting #10 (no NAT) to 172.30.0.188:5060:
NOTIFY sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK404b135e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as057e6e03
To: "4149" <sip:4149@customer.com>;tag=2B89CD1A-E8267AE9
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 4156d055-4140cd6c-fabd396b@172.30.0.188
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4149@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:46] Really destroying SIP dialog '4750d9ac-8361b091-62b430d2@172.30.0.167' Method: REGISTER
[2011-11-03 06:46:46] Really destroying SIP dialog 'c2f6c859-ff2f4a9c-155d1a97@172.30.0.175' Method: REGISTER
[2011-11-03 06:46:46] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 4156d055-4140cd6c-fabd396b@172.30.0.188 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2011-11-03 06:46:46] Retransmitting #10 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:48] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:48] Retransmitting #10 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:48] Really destroying SIP dialog 'c36bdc75-77e4d686-91aca9af@172.30.0.183' Method: REGISTER
[2011-11-03 06:46:48] Retransmitting #10 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:49] Really destroying SIP dialog '9e5886d1-1a3bbf70-bd6245c7@172.30.0.192' Method: REGISTER
[2011-11-03 06:46:49] Retransmitting #10 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:49] Retransmitting #10 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:49] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5cf84e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08f2b599
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f0ee2f145c66737611554f706a6b927@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:50] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fce5058
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7add6ab9
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0f56c67f3176111e752c956db9a647@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:50] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK46a48879
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0e2b6ed3
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 076de64578d1c0ed6f42029c718d2ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:50] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1650e331
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as295d85a9
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7d8202c062de544c552b9fb82ffd662e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:50] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d89f6d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41137a73
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15376a253f572c071b2c10f003325e7b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:50] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK501885d1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71460c09
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0fcc693aebe1f063db9c2023eb2e0d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:50] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5cf84e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08f2b599
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f0ee2f145c66737611554f706a6b927@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Retransmitting #11 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:51] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fce5058
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7add6ab9
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0f56c67f3176111e752c956db9a647@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK46a48879
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0e2b6ed3
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 076de64578d1c0ed6f42029c718d2ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1650e331
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as295d85a9
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7d8202c062de544c552b9fb82ffd662e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37d29bc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as625e7ba4
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 773664ab504d12922047cf597ebffe95@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d89f6d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41137a73
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15376a253f572c071b2c10f003325e7b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28fac45f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as02ac47ff
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 679d65a7596c82dc7fc740200fe2739e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK501885d1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71460c09
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0fcc693aebe1f063db9c2023eb2e0d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:51] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5cf84e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08f2b599
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f0ee2f145c66737611554f706a6b927@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fce5058
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7add6ab9
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0f56c67f3176111e752c956db9a647@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK46a48879
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0e2b6ed3
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 076de64578d1c0ed6f42029c718d2ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:52] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1650e331
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as295d85a9
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7d8202c062de544c552b9fb82ffd662e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #11 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:52] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78e19638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2925a681
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 70cb557923e08c17688ddd32018c05f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37d29bc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as625e7ba4
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 773664ab504d12922047cf597ebffe95@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78908aeb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as164aff54
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a34489f691bfbc52ff8505129cd785e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #11 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:52] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d89f6d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41137a73
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15376a253f572c071b2c10f003325e7b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28fac45f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as02ac47ff
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 679d65a7596c82dc7fc740200fe2739e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ab4556b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as30760cc0
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5bccedb97e37fa251d81fa3d4d6391da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7dd00b66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as433feee9
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d3e37520fde6a7a55c26f5b46362089@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK501885d1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71460c09
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0fcc693aebe1f063db9c2023eb2e0d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:52] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5cf84e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08f2b599
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f0ee2f145c66737611554f706a6b927@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fce5058
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7add6ab9
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0f56c67f3176111e752c956db9a647@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0924136f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as794d73ee
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 45d4b4c059823f8e01ab966278290c7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK632378fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4a2fe689
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fdcfc024751c8a13c6e46480c0ac062@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5387d2b3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ef32bf0
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 781c6d840d0d3e6718c7c95a28ebb85c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK46a48879
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0e2b6ed3
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 076de64578d1c0ed6f42029c718d2ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1650e331
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as295d85a9
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7d8202c062de544c552b9fb82ffd662e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78e19638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2925a681
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 70cb557923e08c17688ddd32018c05f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #11 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:53] Retransmitting #11 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:53] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37d29bc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as625e7ba4
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 773664ab504d12922047cf597ebffe95@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78908aeb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as164aff54
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a34489f691bfbc52ff8505129cd785e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d89f6d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41137a73
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15376a253f572c071b2c10f003325e7b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28fac45f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as02ac47ff
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 679d65a7596c82dc7fc740200fe2739e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ab4556b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as30760cc0
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5bccedb97e37fa251d81fa3d4d6391da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK501885d1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71460c09
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0fcc693aebe1f063db9c2023eb2e0d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f5cf84e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08f2b599
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f0ee2f145c66737611554f706a6b927@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:53] Really destroying SIP dialog '6f0ee2f145c66737611554f706a6b927@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:54] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7dd00b66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as433feee9
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d3e37520fde6a7a55c26f5b46362089@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fce5058
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7add6ab9
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0f56c67f3176111e752c956db9a647@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Really destroying SIP dialog '2a0f56c67f3176111e752c956db9a647@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:54] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0924136f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as794d73ee
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 45d4b4c059823f8e01ab966278290c7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK632378fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4a2fe689
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fdcfc024751c8a13c6e46480c0ac062@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK46a48879
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0e2b6ed3
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 076de64578d1c0ed6f42029c718d2ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Really destroying SIP dialog '076de64578d1c0ed6f42029c718d2ff2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:54] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5387d2b3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ef32bf0
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 781c6d840d0d3e6718c7c95a28ebb85c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1650e331
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as295d85a9
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7d8202c062de544c552b9fb82ffd662e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Really destroying SIP dialog '7d8202c062de544c552b9fb82ffd662e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:54] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78e19638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2925a681
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 70cb557923e08c17688ddd32018c05f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37d29bc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as625e7ba4
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 773664ab504d12922047cf597ebffe95@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78908aeb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as164aff54
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a34489f691bfbc52ff8505129cd785e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5315025e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41d7e74c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 429e1d0021c1a621387b5af941e88920@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1d89f6d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41137a73
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15376a253f572c071b2c10f003325e7b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Really destroying SIP dialog '15376a253f572c071b2c10f003325e7b@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:54] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28fac45f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as02ac47ff
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 679d65a7596c82dc7fc740200fe2739e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ab4556b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as30760cc0
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5bccedb97e37fa251d81fa3d4d6391da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK501885d1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as71460c09
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2a0fcc693aebe1f063db9c2023eb2e0d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:54] Really destroying SIP dialog '2a0fcc693aebe1f063db9c2023eb2e0d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:55] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7dd00b66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as433feee9
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d3e37520fde6a7a55c26f5b46362089@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #12 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:55] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fb65a38
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37a481e6
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 396f93b03694f25e6c45cd8d17e75a8a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4896937e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as681d47b6
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 10047afc1aaf2f67522b22dc5a158fd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0924136f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as794d73ee
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 45d4b4c059823f8e01ab966278290c7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK632378fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4a2fe689
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fdcfc024751c8a13c6e46480c0ac062@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5387d2b3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ef32bf0
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 781c6d840d0d3e6718c7c95a28ebb85c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78e19638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2925a681
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 70cb557923e08c17688ddd32018c05f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37d29bc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as625e7ba4
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 773664ab504d12922047cf597ebffe95@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Really destroying SIP dialog '773664ab504d12922047cf597ebffe95@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:55] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cbd1a96
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2c10dbf3
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53d2ba844375a9325fa0fa291f0a296f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78908aeb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as164aff54
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a34489f691bfbc52ff8505129cd785e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5315025e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41d7e74c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 429e1d0021c1a621387b5af941e88920@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b10954f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e11374e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 06766833592b8f1d72172e7850275496@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK28fac45f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as02ac47ff
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 679d65a7596c82dc7fc740200fe2739e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Really destroying SIP dialog '679d65a7596c82dc7fc740200fe2739e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:55] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ab4556b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as30760cc0
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5bccedb97e37fa251d81fa3d4d6391da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:55] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7dd00b66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as433feee9
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d3e37520fde6a7a55c26f5b46362089@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fb65a38
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37a481e6
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 396f93b03694f25e6c45cd8d17e75a8a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4896937e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as681d47b6
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 10047afc1aaf2f67522b22dc5a158fd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0924136f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as794d73ee
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 45d4b4c059823f8e01ab966278290c7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK632378fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4a2fe689
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fdcfc024751c8a13c6e46480c0ac062@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5387d2b3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ef32bf0
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 781c6d840d0d3e6718c7c95a28ebb85c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:56] Retransmitting #12 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:56] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78e19638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2925a681
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 70cb557923e08c17688ddd32018c05f2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Really destroying SIP dialog '70cb557923e08c17688ddd32018c05f2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:56] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78908aeb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as164aff54
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a34489f691bfbc52ff8505129cd785e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cbd1a96
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2c10dbf3
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53d2ba844375a9325fa0fa291f0a296f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Really destroying SIP dialog '3a34489f691bfbc52ff8505129cd785e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:56] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5315025e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41d7e74c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 429e1d0021c1a621387b5af941e88920@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #12 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:56] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b10954f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e11374e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 06766833592b8f1d72172e7850275496@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ab4556b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as30760cc0
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5bccedb97e37fa251d81fa3d4d6391da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Really destroying SIP dialog '5bccedb97e37fa251d81fa3d4d6391da@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:56] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7dd00b66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as433feee9
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d3e37520fde6a7a55c26f5b46362089@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:56] Really destroying SIP dialog '4d3e37520fde6a7a55c26f5b46362089@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:57] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fb65a38
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37a481e6
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 396f93b03694f25e6c45cd8d17e75a8a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:57] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4896937e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as681d47b6
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 10047afc1aaf2f67522b22dc5a158fd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:57] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0924136f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as794d73ee
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 45d4b4c059823f8e01ab966278290c7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:57] Really destroying SIP dialog '45d4b4c059823f8e01ab966278290c7f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:57] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK632378fb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4a2fe689
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2fdcfc024751c8a13c6e46480c0ac062@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:57] Really destroying SIP dialog '2fdcfc024751c8a13c6e46480c0ac062@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:57] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5387d2b3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ef32bf0
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 781c6d840d0d3e6718c7c95a28ebb85c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:57] Really destroying SIP dialog '781c6d840d0d3e6718c7c95a28ebb85c@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:57] Retransmitting #12 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:57] Retransmitting #12 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:57] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cbd1a96
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2c10dbf3
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53d2ba844375a9325fa0fa291f0a296f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:57] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5315025e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41d7e74c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 429e1d0021c1a621387b5af941e88920@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:57] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b10954f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e11374e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 06766833592b8f1d72172e7850275496@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:58] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fb65a38
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37a481e6
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 396f93b03694f25e6c45cd8d17e75a8a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:58] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4896937e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as681d47b6
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 10047afc1aaf2f67522b22dc5a158fd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:58] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cbd1a96
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2c10dbf3
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53d2ba844375a9325fa0fa291f0a296f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:58] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5315025e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41d7e74c
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 429e1d0021c1a621387b5af941e88920@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:58] Really destroying SIP dialog '429e1d0021c1a621387b5af941e88920@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:58] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b10954f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e11374e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 06766833592b8f1d72172e7850275496@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:58] Retransmitting #13 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK50427db2
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as718dbb37
To: "4123" <sip:4123@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: 3cb94263-cdadaa12-c44b8bf9@172.30.0.184
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:46:59] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fb65a38
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37a481e6
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 396f93b03694f25e6c45cd8d17e75a8a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:59] Really destroying SIP dialog '396f93b03694f25e6c45cd8d17e75a8a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:59] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4896937e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as681d47b6
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 10047afc1aaf2f67522b22dc5a158fd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:59] Really destroying SIP dialog '10047afc1aaf2f67522b22dc5a158fd6@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:59] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4cbd1a96
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2c10dbf3
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53d2ba844375a9325fa0fa291f0a296f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:59] Really destroying SIP dialog '53d2ba844375a9325fa0fa291f0a296f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:46:59] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b10954f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7e11374e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 06766833592b8f1d72172e7850275496@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:46:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:46:59] Really destroying SIP dialog '06766833592b8f1d72172e7850275496@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:00] Retransmitting #7 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:00] Retransmitting #13 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:00] Retransmitting #13 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:01] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3cb94263-cdadaa12-c44b8bf9@172.30.0.184 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 47552ms with no response
[2011-11-03 06:47:01] Retransmitting #13 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:01] Retransmitting #13 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:03] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK522bda69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fb8a466
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ac305227014d65a6dadbb582f5e023a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:04] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18fc5c58
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1afcb965
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4755cd733b1989f61542bd772cd6d63f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:04] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14293619
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as355a7ab8
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e608c1b27f4f7854f4949ce1f13a957@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:04] Retransmitting #8 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:04] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24220491
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as20ab45a5
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53ea7e4762237ebb146bddc6691e93e2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:04] Retransmitting #14 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:04] Retransmitting #14 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK548fc1b9
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as6b236d4d
To: "4148" <sip:4148@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.192
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:04] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK600a8c22
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b5f6d2c
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15196de00671106560931050504c5edb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:04] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b3e2446
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60ce4b3a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7c1080771d7795af1d3c2cc70e73352e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:05] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK522bda69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fb8a466
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ac305227014d65a6dadbb582f5e023a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:05] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18fc5c58
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1afcb965
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4755cd733b1989f61542bd772cd6d63f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:05] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14293619
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as355a7ab8
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e608c1b27f4f7854f4949ce1f13a957@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:05] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24220491
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as20ab45a5
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53ea7e4762237ebb146bddc6691e93e2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:05] Retransmitting #14 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:05] Retransmitting #14 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:05] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18c9c9ee
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4849120e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 46f21e8d580dfe4961d391d8304ce1c0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:05] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK600a8c22
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b5f6d2c
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15196de00671106560931050504c5edb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:05] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7779b026
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25ef318c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 288fa9e15620736945766484086ae167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK522bda69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fb8a466
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ac305227014d65a6dadbb582f5e023a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b3e2446
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60ce4b3a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7c1080771d7795af1d3c2cc70e73352e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18fc5c58
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1afcb965
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4755cd733b1989f61542bd772cd6d63f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14293619
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as355a7ab8
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e608c1b27f4f7854f4949ce1f13a957@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24220491
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as20ab45a5
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53ea7e4762237ebb146bddc6691e93e2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c5af769
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4b5bff3f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62adf85f20d0e373095dc38e64c3f430@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18c9c9ee
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4849120e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 46f21e8d580dfe4961d391d8304ce1c0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b1a2efb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b0f50ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7710811e3687ef756e126c250c9f5310@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK600a8c22
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b5f6d2c
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15196de00671106560931050504c5edb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7779b026
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25ef318c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 288fa9e15620736945766484086ae167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:06] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK522bda69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fb8a466
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ac305227014d65a6dadbb582f5e023a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b3e2446
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60ce4b3a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7c1080771d7795af1d3c2cc70e73352e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b81f859
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5cdfe121
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4cc920de6d7fe64f521f746f6e61aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK239d9f29
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1cf352cf
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c9cd4d972c0e1740de296826f1fe818@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18fc5c58
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1afcb965
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4755cd733b1989f61542bd772cd6d63f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62cd418d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ea532b3
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4e83d3b91b11ccf04b299d6c1240656e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14293619
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as355a7ab8
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e608c1b27f4f7854f4949ce1f13a957@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6838a6c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5629a936
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 201efe2e0a98d4530b942e930e4143b5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76a805f5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as619448cd
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5966c2f4578eaee953e5a3112f014ff7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24220491
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as20ab45a5
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53ea7e4762237ebb146bddc6691e93e2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c5af769
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4b5bff3f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62adf85f20d0e373095dc38e64c3f430@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18c9c9ee
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4849120e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 46f21e8d580dfe4961d391d8304ce1c0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b1a2efb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b0f50ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7710811e3687ef756e126c250c9f5310@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK600a8c22
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b5f6d2c
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15196de00671106560931050504c5edb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7779b026
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25ef318c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 288fa9e15620736945766484086ae167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9e7ede05-62c36944-73d7df3b@172.30.0.192 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 53074ms with no response
[2011-11-03 06:47:08] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK522bda69
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fb8a466
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ac305227014d65a6dadbb582f5e023a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b3e2446
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60ce4b3a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7c1080771d7795af1d3c2cc70e73352e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Really destroying SIP dialog '1ac305227014d65a6dadbb582f5e023a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:08] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b81f859
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5cdfe121
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4cc920de6d7fe64f521f746f6e61aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK239d9f29
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1cf352cf
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c9cd4d972c0e1740de296826f1fe818@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18fc5c58
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1afcb965
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4755cd733b1989f61542bd772cd6d63f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Really destroying SIP dialog '4755cd733b1989f61542bd772cd6d63f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:08] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62cd418d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ea532b3
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4e83d3b91b11ccf04b299d6c1240656e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14293619
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as355a7ab8
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3e608c1b27f4f7854f4949ce1f13a957@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Really destroying SIP dialog '3e608c1b27f4f7854f4949ce1f13a957@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:08] Retransmitting #9 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:08] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6838a6c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5629a936
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 201efe2e0a98d4530b942e930e4143b5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76a805f5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as619448cd
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5966c2f4578eaee953e5a3112f014ff7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #15 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:08] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK24220491
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as20ab45a5
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 53ea7e4762237ebb146bddc6691e93e2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Really destroying SIP dialog '53ea7e4762237ebb146bddc6691e93e2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:08] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c5af769
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4b5bff3f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62adf85f20d0e373095dc38e64c3f430@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18c9c9ee
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4849120e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 46f21e8d580dfe4961d391d8304ce1c0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b1a2efb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b0f50ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7710811e3687ef756e126c250c9f5310@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK488f3575
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as071f19c7
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 18edad7568823ffa7e62c4304c2af0e5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK600a8c22
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b5f6d2c
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15196de00671106560931050504c5edb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Really destroying SIP dialog '15196de00671106560931050504c5edb@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:08] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7779b026
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25ef318c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 288fa9e15620736945766484086ae167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:08] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b3e2446
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60ce4b3a
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7c1080771d7795af1d3c2cc70e73352e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Really destroying SIP dialog '7c1080771d7795af1d3c2cc70e73352e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:09] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b81f859
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5cdfe121
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4cc920de6d7fe64f521f746f6e61aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK239d9f29
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1cf352cf
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c9cd4d972c0e1740de296826f1fe818@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK316888a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as54620926
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5ce1062654eb32639daa4e2e5bce8c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ac6d520
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3ee41a50
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2212b96a1fa125167ebf96667c4c7a5d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62cd418d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ea532b3
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4e83d3b91b11ccf04b299d6c1240656e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6838a6c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5629a936
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 201efe2e0a98d4530b942e930e4143b5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76a805f5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as619448cd
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5966c2f4578eaee953e5a3112f014ff7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #15 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:09] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c5af769
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4b5bff3f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62adf85f20d0e373095dc38e64c3f430@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #15 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:09] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK18c9c9ee
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4849120e
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 46f21e8d580dfe4961d391d8304ce1c0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Really destroying SIP dialog '46f21e8d580dfe4961d391d8304ce1c0@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:09] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b1a2efb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b0f50ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7710811e3687ef756e126c250c9f5310@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59454ac5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3a280d07
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b03ac0f37405d7162cdef8c27e020b7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK488f3575
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as071f19c7
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 18edad7568823ffa7e62c4304c2af0e5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f66af2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as640219f6
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0507d01609da320672db2bbd01192e76@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7779b026
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25ef318c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 288fa9e15620736945766484086ae167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:09] Really destroying SIP dialog '288fa9e15620736945766484086ae167@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:10] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b81f859
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5cdfe121
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4cc920de6d7fe64f521f746f6e61aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK239d9f29
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1cf352cf
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c9cd4d972c0e1740de296826f1fe818@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK316888a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as54620926
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5ce1062654eb32639daa4e2e5bce8c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ac6d520
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3ee41a50
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2212b96a1fa125167ebf96667c4c7a5d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62cd418d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ea532b3
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4e83d3b91b11ccf04b299d6c1240656e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6838a6c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5629a936
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 201efe2e0a98d4530b942e930e4143b5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76a805f5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as619448cd
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5966c2f4578eaee953e5a3112f014ff7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c5af769
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4b5bff3f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62adf85f20d0e373095dc38e64c3f430@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Really destroying SIP dialog '62adf85f20d0e373095dc38e64c3f430@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:10] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b1a2efb
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7b0f50ef
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7710811e3687ef756e126c250c9f5310@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Really destroying SIP dialog '7710811e3687ef756e126c250c9f5310@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:10] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59454ac5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3a280d07
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b03ac0f37405d7162cdef8c27e020b7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK488f3575
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as071f19c7
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 18edad7568823ffa7e62c4304c2af0e5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:10] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f66af2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as640219f6
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0507d01609da320672db2bbd01192e76@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b81f859
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5cdfe121
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4cc920de6d7fe64f521f746f6e61aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Really destroying SIP dialog '4cc920de6d7fe64f521f746f6e61aee3@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:11] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK239d9f29
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1cf352cf
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4c9cd4d972c0e1740de296826f1fe818@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Really destroying SIP dialog '4c9cd4d972c0e1740de296826f1fe818@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:11] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK316888a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as54620926
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5ce1062654eb32639daa4e2e5bce8c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ac6d520
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3ee41a50
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2212b96a1fa125167ebf96667c4c7a5d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11]
<--- SIP read from UDP:172.30.0.189:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK86de60c05FF4BFBB
From: "Boardroom" <sip:4109@customer.com>;tag=22328416-2F361C19
To: <sip:4109@customer.com>
CSeq: 1 REGISTER
Call-ID: 4814d2fa-ee5ed3ad-4e3bf144@172.30.0.189
Contact: <sip:4109@172.30.0.189:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:11] --- (12 headers 0 lines) ---
[2011-11-03 06:47:11] Sending to 172.30.0.189:5060 (no NAT)
[2011-11-03 06:47:11]
<--- Transmitting (no NAT) to 172.30.0.189:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK86de60c05FF4BFBB;received=172.30.0.189
From: "Boardroom" <sip:4109@customer.com>;tag=22328416-2F361C19
To: <sip:4109@customer.com>;tag=as3e115581
Call-ID: 4814d2fa-ee5ed3ad-4e3bf144@172.30.0.189
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c6c0152"
Content-Length: 0


<------------>
[2011-11-03 06:47:11] Scheduling destruction of SIP dialog '4814d2fa-ee5ed3ad-4e3bf144@172.30.0.189' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:11] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62cd418d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5ea532b3
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4e83d3b91b11ccf04b299d6c1240656e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Really destroying SIP dialog '4e83d3b91b11ccf04b299d6c1240656e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:11]
<--- SIP read from UDP:172.30.0.189:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK92413c6f9F5E161E
From: "Boardroom" <sip:4109@customer.com>;tag=22328416-2F361C19
To: <sip:4109@customer.com>
CSeq: 2 REGISTER
Call-ID: 4814d2fa-ee5ed3ad-4e3bf144@172.30.0.189
Contact: <sip:4109@172.30.0.189:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4109", realm="asterisk", nonce="3c6c0152", uri="sip:customer.com:5060", response="0ec8c72340a85dc95cb735f0caaa9a38", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:11] --- (13 headers 0 lines) ---
[2011-11-03 06:47:11] Sending to 172.30.0.189:5060 (no NAT)
[2011-11-03 06:47:11]     -- Registered SIP '4109' at 172.30.0.189:5060
[2011-11-03 06:47:11] Reliably Transmitting (no NAT) to 172.30.0.189:5060:
OPTIONS sip:4109@172.30.0.189:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4152fad6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as45ef9e57
To: <sip:4109@172.30.0.189:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0bf8965b5fefa4a37254fb3449cb2d85@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11]
<--- Transmitting (no NAT) to 172.30.0.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK92413c6f9F5E161E;received=172.30.0.189
From: "Boardroom" <sip:4109@customer.com>;tag=22328416-2F361C19
To: <sip:4109@customer.com>;tag=as3e115581
Call-ID: 4814d2fa-ee5ed3ad-4e3bf144@172.30.0.189
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.189:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:11 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:11] Scheduling destruction of SIP dialog '4814d2fa-ee5ed3ad-4e3bf144@172.30.0.189' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:11] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6838a6c6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5629a936
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 201efe2e0a98d4530b942e930e4143b5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Really destroying SIP dialog '201efe2e0a98d4530b942e930e4143b5@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:11] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76a805f5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as619448cd
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5966c2f4578eaee953e5a3112f014ff7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Really destroying SIP dialog '5966c2f4578eaee953e5a3112f014ff7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:11]
<--- SIP read from UDP:172.30.0.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4152fad6
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as45ef9e57
To: "Boardroom" <sip:4109@172.30.0.189:5060>;tag=8E128008-88F60263
CSeq: 102 OPTIONS
Call-ID: 0bf8965b5fefa4a37254fb3449cb2d85@172.30.0.10:5060
Contact: <sip:4109@172.30.0.189:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:11] --- (14 headers 0 lines) ---
[2011-11-03 06:47:11] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4109' is now Reachable. (72ms / 2000ms)
[2011-11-03 06:47:11] Really destroying SIP dialog '0bf8965b5fefa4a37254fb3449cb2d85@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4113 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4116 (queued)
[2011-11-03 06:47:11] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:47:11] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:47:11] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4148
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4125 (queued)
[2011-11-03 06:47:11] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:47:11] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:47:11] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4135
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4137 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4134
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4138
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4128 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:11] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:47:11] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:47:11] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2667fd0e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:11]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2667fd0e
From: <sip:4109@customer.com>;tag=as0b69ee13
To: "4136" <sip:4136@customer.com>;tag=C5BC4CA-8E22E023
CSeq: 105 NOTIFY
Call-ID: e14f6bb3-4c9311cc-28e5e6a5@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:11] --- (11 headers 0 lines) ---
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4136
[2011-11-03 06:47:11] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:11] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59454ac5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3a280d07
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b03ac0f37405d7162cdef8c27e020b7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:47:11]   == Extension Changed 4109[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:11] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK488f3575
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as071f19c7
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 18edad7568823ffa7e62c4304c2af0e5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:11] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f66af2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as640219f6
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0507d01609da320672db2bbd01192e76@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:12] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK316888a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as54620926
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5ce1062654eb32639daa4e2e5bce8c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:12] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ac6d520
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3ee41a50
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2212b96a1fa125167ebf96667c4c7a5d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:12]
<--- SIP read from UDP:172.30.0.189:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bKeac48597D7584926
From: "Boardroom" <sip:4109@customer.com>;tag=1CEAEC95-F4EA510C
To: <sip:4109@customer.com:5060>
CSeq: 1 SUBSCRIBE
Call-ID: 9d1e35e9-ba65e850-62137a0b@172.30.0.189
Contact: <sip:4109@172.30.0.189:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:12] --- (15 headers 0 lines) ---
[2011-11-03 06:47:12] Creating new subscription
[2011-11-03 06:47:12] Sending to 172.30.0.189:5060 (no NAT)
[2011-11-03 06:47:12] list_route: hop: <sip:4109@172.30.0.189:5060>
[2011-11-03 06:47:12] Found peer '4109' for '4109' from 172.30.0.189:5060
[2011-11-03 06:47:12]
<--- Transmitting (no NAT) to 172.30.0.189:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bKeac48597D7584926;received=172.30.0.189
From: "Boardroom" <sip:4109@customer.com>;tag=1CEAEC95-F4EA510C
To: <sip:4109@customer.com:5060>;tag=as4a6e13cf
Call-ID: 9d1e35e9-ba65e850-62137a0b@172.30.0.189
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0108b95d"
Content-Length: 0


<------------>
[2011-11-03 06:47:12] Scheduling destruction of SIP dialog '9d1e35e9-ba65e850-62137a0b@172.30.0.189' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:12] Retransmitting #10 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09d1760f
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as7ddb3c83
To: "4118" <sip:4118@customer.com>;tag=878ABE7E-3CA31493
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: a8664db1-74778406-262df01b@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:12]
<--- SIP read from UDP:172.30.0.189:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK843f410aC619027D
From: "Boardroom" <sip:4109@customer.com>;tag=1CEAEC95-F4EA510C
To: <sip:4109@customer.com:5060>
CSeq: 2 SUBSCRIBE
Call-ID: 9d1e35e9-ba65e850-62137a0b@172.30.0.189
Contact: <sip:4109@172.30.0.189:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Authorization: Digest username="4109", realm="asterisk", nonce="0108b95d", uri="sip:4109@customer.com:5060", response="6a16bdf23ffdd1de856fe8806cb2b240", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:12] --- (16 headers 0 lines) ---
[2011-11-03 06:47:12] Creating new subscription
[2011-11-03 06:47:12] Sending to 172.30.0.189:5060 (no NAT)
[2011-11-03 06:47:12] Found peer '4109' for '4109' from 172.30.0.189:5060
[2011-11-03 06:47:12] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:12]
<--- Transmitting (no NAT) to 172.30.0.189:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK843f410aC619027D;received=172.30.0.189
From: "Boardroom" <sip:4109@customer.com>;tag=1CEAEC95-F4EA510C
To: <sip:4109@customer.com:5060>;tag=as4a6e13cf
Call-ID: 9d1e35e9-ba65e850-62137a0b@172.30.0.189
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:47:12] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:12] Retransmitting #16 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a34d7c6
Max-Forwards: 70
From: <sip:4136@customer.com>;tag=as5bd12756
To: "4173" <sip:4173@customer.com>;tag=C7DBCA49-A00684A8
Contact: <sip:4136@172.30.0.10:5060>
Call-ID: c44b8bf9-ba630fd8-6127cb6f@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4136">
<address uri="sip:4136@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:12] Really destroying SIP dialog '9d1e35e9-ba65e850-62137a0b@172.30.0.189' Method: SUBSCRIBE
[2011-11-03 06:47:12] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59454ac5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3a280d07
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b03ac0f37405d7162cdef8c27e020b7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a8664db1-74778406-262df01b@172.30.0.173 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31986ms with no response
[2011-11-03 06:47:12] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK488f3575
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as071f19c7
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 18edad7568823ffa7e62c4304c2af0e5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:12] Really destroying SIP dialog '18edad7568823ffa7e62c4304c2af0e5@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:12] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f66af2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as640219f6
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0507d01609da320672db2bbd01192e76@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:12] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission c44b8bf9-ba630fd8-6127cb6f@172.30.0.178 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 59126ms with no response
[2011-11-03 06:47:13] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK316888a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as54620926
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1d5ce1062654eb32639daa4e2e5bce8c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:13] Really destroying SIP dialog '1d5ce1062654eb32639daa4e2e5bce8c@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:13] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ac6d520
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3ee41a50
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2212b96a1fa125167ebf96667c4c7a5d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:13] Really destroying SIP dialog '2212b96a1fa125167ebf96667c4c7a5d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:13] Retransmitting #16 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b7c446a
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as4f0404d9
To: "4173" <sip:4173@customer.com>;tag=C0643B01-E01DF120
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 9e7ede05-62c36944-73d7df3b@172.30.0.178
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:13] Retransmitting #16 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK496fbab9
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as34da11d0
To: "4173" <sip:4173@customer.com>;tag=9D60189D-33E2869C
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:13] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59454ac5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3a280d07
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b03ac0f37405d7162cdef8c27e020b7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:13] Really destroying SIP dialog '6b03ac0f37405d7162cdef8c27e020b7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:13] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25f66af2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as640219f6
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0507d01609da320672db2bbd01192e76@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:13] Really destroying SIP dialog '0507d01609da320672db2bbd01192e76@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:13] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:14] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9e7ede05-62c36944-73d7df3b@172.30.0.178 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 59136ms with no response
[2011-11-03 06:47:14] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3c6ccdbd-7b4d0cbc-9c140473@172.30.0.178 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 59137ms with no response
[2011-11-03 06:47:15] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:16]
<--- SIP read from UDP:172.30.0.190:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bK1aa27aad2220A43C
From: "Meeting Room" <sip:4130@customer.com>;tag=65975D1F-98D6854E
To: <sip:4130@customer.com>
CSeq: 1 REGISTER
Call-ID: bf918fb-34b59fea-1750b009@172.30.0.190
Contact: <sip:4130@172.30.0.190:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:16] --- (12 headers 0 lines) ---
[2011-11-03 06:47:16] Sending to 172.30.0.190:5060 (no NAT)
[2011-11-03 06:47:16]
<--- Transmitting (no NAT) to 172.30.0.190:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bK1aa27aad2220A43C;received=172.30.0.190
From: "Meeting Room" <sip:4130@customer.com>;tag=65975D1F-98D6854E
To: <sip:4130@customer.com>;tag=as6813ddd0
Call-ID: bf918fb-34b59fea-1750b009@172.30.0.190
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="512bc344"
Content-Length: 0


<------------>
[2011-11-03 06:47:16] Scheduling destruction of SIP dialog 'bf918fb-34b59fea-1750b009@172.30.0.190' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:16]
<--- SIP read from UDP:172.30.0.190:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bKc6f470588AE9B7D7
From: "Meeting Room" <sip:4130@customer.com>;tag=65975D1F-98D6854E
To: <sip:4130@customer.com>
CSeq: 2 REGISTER
Call-ID: bf918fb-34b59fea-1750b009@172.30.0.190
Contact: <sip:4130@172.30.0.190:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4130", realm="asterisk", nonce="512bc344", uri="sip:customer.com:5060", response="699b1cfc220be00a03c8e11d36e643c6", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:16] --- (13 headers 0 lines) ---
[2011-11-03 06:47:16] Sending to 172.30.0.190:5060 (no NAT)
[2011-11-03 06:47:16]     -- Registered SIP '4130' at 172.30.0.190:5060
[2011-11-03 06:47:16] Reliably Transmitting (no NAT) to 172.30.0.190:5060:
OPTIONS sip:4130@172.30.0.190:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78ed38e0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as47205832
To: <sip:4130@172.30.0.190:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 66b1466330717bef69fe773737e32592@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:16]
<--- Transmitting (no NAT) to 172.30.0.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bKc6f470588AE9B7D7;received=172.30.0.190
From: "Meeting Room" <sip:4130@customer.com>;tag=65975D1F-98D6854E
To: <sip:4130@customer.com>;tag=as6813ddd0
Call-ID: bf918fb-34b59fea-1750b009@172.30.0.190
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4130@172.30.0.190:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:16 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:16] Scheduling destruction of SIP dialog 'bf918fb-34b59fea-1750b009@172.30.0.190' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4131 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4138 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4128 (queued)
[2011-11-03 06:47:16] set_destination: Parsing <sip:4125@172.30.0.168:5060> for address/port to send to
[2011-11-03 06:47:16] set_destination: set destination to 172.30.0.168:5060
[2011-11-03 06:47:16] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c88042b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4125
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:16]   == Extension Changed 4130[default-local] new state Idle for Notify User 4116 (queued)
[2011-11-03 06:47:16]
<--- SIP read from UDP:172.30.0.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78ed38e0
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as47205832
To: "Meeting Room" <sip:4130@172.30.0.190:5060>;tag=706FA865-9B326F74
CSeq: 102 OPTIONS
Call-ID: 66b1466330717bef69fe773737e32592@172.30.0.10:5060
Contact: <sip:4130@172.30.0.190:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:16] --- (14 headers 0 lines) ---
[2011-11-03 06:47:16] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4130' is now Reachable. (80ms / 2000ms)
[2011-11-03 06:47:16] Really destroying SIP dialog '66b1466330717bef69fe773737e32592@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:16] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c88042b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:17] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c88042b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:17] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:17] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c88042b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:17]
<--- SIP read from UDP:172.30.0.190:5060 --->
SUBSCRIBE sip:4130@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bK8f2e61909CE69E8F
From: "Meeting Room" <sip:4130@customer.com>;tag=2FD68E22-B64723C1
To: <sip:4130@customer.com:5060>
CSeq: 1 SUBSCRIBE
Call-ID: 6bfe1be-2bcee21d-a4ec06ac@172.30.0.190
Contact: <sip:4130@172.30.0.190:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:17] --- (15 headers 0 lines) ---
[2011-11-03 06:47:17] Creating new subscription
[2011-11-03 06:47:17] Sending to 172.30.0.190:5060 (no NAT)
[2011-11-03 06:47:17] list_route: hop: <sip:4130@172.30.0.190:5060>
[2011-11-03 06:47:17] Found peer '4130' for '4130' from 172.30.0.190:5060
[2011-11-03 06:47:17]
<--- Transmitting (no NAT) to 172.30.0.190:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bK8f2e61909CE69E8F;received=172.30.0.190
From: "Meeting Room" <sip:4130@customer.com>;tag=2FD68E22-B64723C1
To: <sip:4130@customer.com:5060>;tag=as212b1f69
Call-ID: 6bfe1be-2bcee21d-a4ec06ac@172.30.0.190
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69ccc72b"
Content-Length: 0


<------------>
[2011-11-03 06:47:17] Scheduling destruction of SIP dialog '6bfe1be-2bcee21d-a4ec06ac@172.30.0.190' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:17]
<--- SIP read from UDP:172.30.0.190:5060 --->
SUBSCRIBE sip:4130@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bK5ce2666b75CEC85A
From: "Meeting Room" <sip:4130@customer.com>;tag=2FD68E22-B64723C1
To: <sip:4130@customer.com:5060>
CSeq: 2 SUBSCRIBE
Call-ID: 6bfe1be-2bcee21d-a4ec06ac@172.30.0.190
Contact: <sip:4130@172.30.0.190:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Authorization: Digest username="4130", realm="asterisk", nonce="69ccc72b", uri="sip:4130@customer.com:5060", response="ea01618f801208d0ebc89bdf2a53dc2f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:17] --- (16 headers 0 lines) ---
[2011-11-03 06:47:17] Creating new subscription
[2011-11-03 06:47:17] Sending to 172.30.0.190:5060 (no NAT)
[2011-11-03 06:47:17] Found peer '4130' for '4130' from 172.30.0.190:5060
[2011-11-03 06:47:17] Looking for 4130 in default-local (domain customer.com:5060)
[2011-11-03 06:47:17]
<--- Transmitting (no NAT) to 172.30.0.190:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bK5ce2666b75CEC85A;received=172.30.0.190
From: "Meeting Room" <sip:4130@customer.com>;tag=2FD68E22-B64723C1
To: <sip:4130@customer.com:5060>;tag=as212b1f69
Call-ID: 6bfe1be-2bcee21d-a4ec06ac@172.30.0.190
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:47:17] Really destroying SIP dialog '6bfe1be-2bcee21d-a4ec06ac@172.30.0.190' Method: SUBSCRIBE
[2011-11-03 06:47:18] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00824b79
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as33c2dadb
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4578625c0a78de0755e2371227044ead@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:18] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK177aa369
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5afc6a59
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 660db88c714ec9e32e173e9029ccb8e0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:18] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26dd51cd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a351821
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6bac251d1436dcf34a24c8c473951615@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:18] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c88042b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:18] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62e38d7d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as703e8551
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c72da2679bf042c4bf491360d01b226@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:18] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5511b5e9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25eadd12
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49388bdc5648a1b71f8355b15dba7e08@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:18]
<--- SIP read from UDP:172.30.0.195:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKec3ce5aa6D9ED4F7
From: "Office 131" <sip:4113@customer.com>;tag=847E92DC-E01A3E91
To: <sip:4113@customer.com>
CSeq: 1 REGISTER
Call-ID: 15b9cd88-ef829ad-97c025f6@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:18] --- (12 headers 0 lines) ---
[2011-11-03 06:47:18] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:18]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKec3ce5aa6D9ED4F7;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=847E92DC-E01A3E91
To: <sip:4113@customer.com>;tag=as26c89ec8
Call-ID: 15b9cd88-ef829ad-97c025f6@172.30.0.195
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c5b6c18"
Content-Length: 0


<------------>
[2011-11-03 06:47:18] Scheduling destruction of SIP dialog '15b9cd88-ef829ad-97c025f6@172.30.0.195' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:18]
<--- SIP read from UDP:172.30.0.195:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKa84b8a82CB5870AF
From: "Office 131" <sip:4113@customer.com>;tag=847E92DC-E01A3E91
To: <sip:4113@customer.com>
CSeq: 2 REGISTER
Call-ID: 15b9cd88-ef829ad-97c025f6@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4113", realm="asterisk", nonce="4c5b6c18", uri="sip:customer.com:5060", response="e400667ad38828591c9682aeded5c823", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:18] --- (13 headers 0 lines) ---
[2011-11-03 06:47:18] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:18]     -- Registered SIP '4113' at 172.30.0.195:5060
[2011-11-03 06:47:18] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
OPTIONS sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK766de783
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5598a1d0
To: <sip:4113@172.30.0.195:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 33ab7a1a32da71f8751bf4906f654102@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:18]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKa84b8a82CB5870AF;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=847E92DC-E01A3E91
To: <sip:4113@customer.com>;tag=as26c89ec8
Call-ID: 15b9cd88-ef829ad-97c025f6@172.30.0.195
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4113@172.30.0.195:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:18 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:18] Scheduling destruction of SIP dialog '15b9cd88-ef829ad-97c025f6@172.30.0.195' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:18] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:47:18] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:47:18] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4141
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:47:18]   == Extension Changed 4113[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:18]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK766de783
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5598a1d0
To: "Office 131" <sip:4113@172.30.0.195:5060>;tag=5C42E3BE-D6B7C19B
CSeq: 102 OPTIONS
Call-ID: 33ab7a1a32da71f8751bf4906f654102@172.30.0.10:5060
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:18] --- (14 headers 0 lines) ---
[2011-11-03 06:47:18] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4113' is now Reachable. (79ms / 2000ms)
[2011-11-03 06:47:19] Really destroying SIP dialog '33ab7a1a32da71f8751bf4906f654102@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:19] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00824b79
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as33c2dadb
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4578625c0a78de0755e2371227044ead@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:19] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK640fd1d2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as751be278
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5563252a4c04f84d096e42394aeecc7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK177aa369
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5afc6a59
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 660db88c714ec9e32e173e9029ccb8e0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:19]
<--- SIP read from UDP:172.30.0.189:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK7badb9984BCFC6B3
From: "Boardroom" <sip:4109@customer.com>;tag=54013D2E-488F6651
To: <sip:4109@customer.com:5060>
CSeq: 1 SUBSCRIBE
Call-ID: 5817e192-4484b565-cba63b9c@172.30.0.189
Contact: <sip:4109@172.30.0.189:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:19] --- (15 headers 0 lines) ---
[2011-11-03 06:47:19] Creating new subscription
[2011-11-03 06:47:19] Sending to 172.30.0.189:5060 (no NAT)
[2011-11-03 06:47:19] list_route: hop: <sip:4109@172.30.0.189:5060>
[2011-11-03 06:47:19] Found peer '4109' for '4109' from 172.30.0.189:5060
[2011-11-03 06:47:19]
<--- Transmitting (no NAT) to 172.30.0.189:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK7badb9984BCFC6B3;received=172.30.0.189
From: "Boardroom" <sip:4109@customer.com>;tag=54013D2E-488F6651
To: <sip:4109@customer.com:5060>;tag=as43ccd97a
Call-ID: 5817e192-4484b565-cba63b9c@172.30.0.189
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6823ad6a"
Content-Length: 0


<------------>
[2011-11-03 06:47:19] Scheduling destruction of SIP dialog '5817e192-4484b565-cba63b9c@172.30.0.189' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:19] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26dd51cd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a351821
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6bac251d1436dcf34a24c8c473951615@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19]
<--- SIP read from UDP:172.30.0.189:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK6ba8c6e76CBD1236
From: "Boardroom" <sip:4109@customer.com>;tag=54013D2E-488F6651
To: <sip:4109@customer.com:5060>
CSeq: 2 SUBSCRIBE
Call-ID: 5817e192-4484b565-cba63b9c@172.30.0.189
Contact: <sip:4109@172.30.0.189:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Authorization: Digest username="4109", realm="asterisk", nonce="6823ad6a", uri="sip:4109@customer.com:5060", response="f06fbba262d284a9e4538b1eb34760a3", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:19] --- (16 headers 0 lines) ---
[2011-11-03 06:47:19] Creating new subscription
[2011-11-03 06:47:19] Sending to 172.30.0.189:5060 (no NAT)
[2011-11-03 06:47:19] Found peer '4109' for '4109' from 172.30.0.189:5060
[2011-11-03 06:47:19] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:19]
<--- Transmitting (no NAT) to 172.30.0.189:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK6ba8c6e76CBD1236;received=172.30.0.189
From: "Boardroom" <sip:4109@customer.com>;tag=54013D2E-488F6651
To: <sip:4109@customer.com:5060>;tag=as43ccd97a
Call-ID: 5817e192-4484b565-cba63b9c@172.30.0.189
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2011-11-03 06:47:19] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:19] Really destroying SIP dialog '5817e192-4484b565-cba63b9c@172.30.0.189' Method: SUBSCRIBE
[2011-11-03 06:47:19] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62e38d7d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as703e8551
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c72da2679bf042c4bf491360d01b226@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK461837fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0ac7f097
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0fc14b0f575ab173437a6b9c7191bb4d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:19] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5511b5e9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25eadd12
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49388bdc5648a1b71f8355b15dba7e08@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK96672ffcB1ECEA31
From: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:19] --- (15 headers 0 lines) ---
[2011-11-03 06:47:19] Creating new subscription
[2011-11-03 06:47:19] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:19] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:19] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:19]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK96672ffcB1ECEA31;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
To: <sip:4119@customer.com>;tag=as68f65716
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f720636"
Content-Length: 0


<------------>
[2011-11-03 06:47:19] Scheduling destruction of SIP dialog 'd62861ca-ab518b97-a964f0a8@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:19]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKafd9534d36537816
From: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
To: <sip:4119@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="6f720636", uri="sip:4119@customer.com:5060", response="6c39fcf33776728a205c786335896ade", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:19] --- (16 headers 0 lines) ---
[2011-11-03 06:47:19] Creating new subscription
[2011-11-03 06:47:19] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:19] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:19] Looking for 4119 in default-local (domain customer.com:5060)
[2011-11-03 06:47:19] Scheduling destruction of SIP dialog 'd62861ca-ab518b97-a964f0a8@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:19]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKafd9534d36537816;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
To: <sip:4119@customer.com>;tag=as68f65716
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4119@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:19] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:19] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:19] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK094c2114
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:19] Retransmitting #5 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c88042b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:19] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK212afd08
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b7e9322
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7973f4bf5410cb832091ce2236afd267@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:19]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK094c2114
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
CSeq: 102 NOTIFY
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:19] --- (11 headers 0 lines) ---
[2011-11-03 06:47:19] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:20] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00824b79
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as33c2dadb
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4578625c0a78de0755e2371227044ead@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK640fd1d2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as751be278
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5563252a4c04f84d096e42394aeecc7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK177aa369
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5afc6a59
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 660db88c714ec9e32e173e9029ccb8e0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20]
<--- SIP read from UDP:172.30.0.167:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK6b6afdd66415B767
From: "4145" <sip:4145@customer.com>;tag=C2F8F338-282FE321
To: <sip:4145@customer.com>
CSeq: 1 REGISTER
Call-ID: ad1a82a4-d535aa1d-aa83522@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:20] --- (12 headers 0 lines) ---
[2011-11-03 06:47:20] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:20]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK6b6afdd66415B767;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=C2F8F338-282FE321
To: <sip:4145@customer.com>;tag=as6b716342
Call-ID: ad1a82a4-d535aa1d-aa83522@172.30.0.167
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="270e02f6"
Content-Length: 0


<------------>
[2011-11-03 06:47:20] Scheduling destruction of SIP dialog 'ad1a82a4-d535aa1d-aa83522@172.30.0.167' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:20] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26dd51cd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a351821
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6bac251d1436dcf34a24c8c473951615@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20]
<--- SIP read from UDP:172.30.0.167:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK52d5e8c3E8C34D0
From: "4145" <sip:4145@customer.com>;tag=C2F8F338-282FE321
To: <sip:4145@customer.com>
CSeq: 2 REGISTER
Call-ID: ad1a82a4-d535aa1d-aa83522@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4145", realm="asterisk", nonce="270e02f6", uri="sip:customer.com:5060", response="d4396288b7d5862e98e00f63c5fd664c", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:20] --- (13 headers 0 lines) ---
[2011-11-03 06:47:20] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:20] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
OPTIONS sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4bafb6a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as39745f17
To: <sip:4145@172.30.0.167:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0d9cf9592e74315b39988b711f1bb6fb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK52d5e8c3E8C34D0;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=C2F8F338-282FE321
To: <sip:4145@customer.com>;tag=as6b716342
Call-ID: ad1a82a4-d535aa1d-aa83522@172.30.0.167
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4145@172.30.0.167:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:20 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:20] Scheduling destruction of SIP dialog 'ad1a82a4-d535aa1d-aa83522@172.30.0.167' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:20]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4bafb6a0
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as39745f17
To: "4145" <sip:4145@172.30.0.167:5060>;tag=183C7DC4-E18B26BD
CSeq: 102 OPTIONS
Call-ID: 0d9cf9592e74315b39988b711f1bb6fb@172.30.0.10:5060
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:20] --- (14 headers 0 lines) ---
[2011-11-03 06:47:20] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4145' is now Reachable. (62ms / 2000ms)
[2011-11-03 06:47:20] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62e38d7d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as703e8551
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c72da2679bf042c4bf491360d01b226@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4113 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4138 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:20]   == Extension Changed 4145[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:20] Really destroying SIP dialog '0d9cf9592e74315b39988b711f1bb6fb@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:20] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e2d5f16
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3568903f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 555690ff22901feb58d5ac8a193a199f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:20] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK461837fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0ac7f097
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0fc14b0f575ab173437a6b9c7191bb4d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5511b5e9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25eadd12
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49388bdc5648a1b71f8355b15dba7e08@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:20]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4149@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKce8be2d3F4DA9794
From: "Office 131" <sip:4113@customer.com>;tag=942FB5A5-666CFF4E
To: <sip:4149@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d0322fa9-86a1d2a2-f670e34f@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:20] --- (15 headers 0 lines) ---
[2011-11-03 06:47:20] Creating new subscription
[2011-11-03 06:47:20] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:20] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:20] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:20]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKce8be2d3F4DA9794;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=942FB5A5-666CFF4E
To: <sip:4149@customer.com>;tag=as6818db20
Call-ID: d0322fa9-86a1d2a2-f670e34f@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="298743e0"
Content-Length: 0


<------------>
[2011-11-03 06:47:20] Scheduling destruction of SIP dialog 'd0322fa9-86a1d2a2-f670e34f@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:20]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4149@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKaa5370c01627BB45
From: "Office 131" <sip:4113@customer.com>;tag=942FB5A5-666CFF4E
To: <sip:4149@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: d0322fa9-86a1d2a2-f670e34f@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="298743e0", uri="sip:4149@customer.com:5060", response="5f7b6197cf744d38d3d4ea916464b4f9", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:20] --- (16 headers 0 lines) ---
[2011-11-03 06:47:20] Creating new subscription
[2011-11-03 06:47:20] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:20] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:20] Looking for 4149 in default-local (domain customer.com:5060)
[2011-11-03 06:47:20] Scheduling destruction of SIP dialog 'd0322fa9-86a1d2a2-f670e34f@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:20]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKaa5370c01627BB45;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=942FB5A5-666CFF4E
To: <sip:4149@customer.com>;tag=as6818db20
Call-ID: d0322fa9-86a1d2a2-f670e34f@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4149@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:20] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:20] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:20] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b447e22
Max-Forwards: 70
From: <sip:4149@customer.com>;tag=as6818db20
To: "Office 131" <sip:4113@customer.com>;tag=942FB5A5-666CFF4E
Contact: <sip:4149@172.30.0.10:5060>
Call-ID: d0322fa9-86a1d2a2-f670e34f@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4149">
<address uri="sip:4149@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:20]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b447e22
From: <sip:4149@customer.com>;tag=as6818db20
To: "Office 131" <sip:4113@customer.com>;tag=942FB5A5-666CFF4E
CSeq: 102 NOTIFY
Call-ID: d0322fa9-86a1d2a2-f670e34f@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:20] --- (11 headers 0 lines) ---
[2011-11-03 06:47:20] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:20] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK212afd08
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b7e9322
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7973f4bf5410cb832091ce2236afd267@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00824b79
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as33c2dadb
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4578625c0a78de0755e2371227044ead@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK640fd1d2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as751be278
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5563252a4c04f84d096e42394aeecc7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK17ad3e46
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as244ea810
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28382b380911cff206cf607d79d861f3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f13ea27
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2eaf1dbe
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2b9cf7385f382b0b1f028d11307ecc5a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK177aa369
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5afc6a59
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 660db88c714ec9e32e173e9029ccb8e0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKde2c9542C542E663
From: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:21] --- (15 headers 0 lines) ---
[2011-11-03 06:47:21] Creating new subscription
[2011-11-03 06:47:21] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:21] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:21] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:21]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKde2c9542C542E663;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
To: <sip:4148@customer.com>;tag=as1c4480b3
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="662c170f"
Content-Length: 0


<------------>
[2011-11-03 06:47:21] Scheduling destruction of SIP dialog 'ecd301f0-4e178379-e5426f4e@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:21] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26dd51cd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a351821
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6bac251d1436dcf34a24c8c473951615@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d3f4e82
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10395db6
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71c4ba862d920c7d1d92ae0d01fbec3d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:21]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK4ad1c37f9DD9B4DC
From: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="662c170f", uri="sip:4148@customer.com:5060", response="8a0b43617641eac71a0fa20566dc18dc", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:21] --- (16 headers 0 lines) ---
[2011-11-03 06:47:21] Creating new subscription
[2011-11-03 06:47:21] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:21] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:21] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:21] Scheduling destruction of SIP dialog 'ecd301f0-4e178379-e5426f4e@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:21]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK4ad1c37f9DD9B4DC;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
To: <sip:4148@customer.com>;tag=as1c4480b3
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:21] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:21] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:21] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK464c8fdc
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:21] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK425cf67f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08e41776
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 68b9769f79fa6bef6cb884ca32af0104@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f0fd860
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as187410ad
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0e44634861a72fd41644f8cc292e9e97@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62e38d7d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as703e8551
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c72da2679bf042c4bf491360d01b226@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK464c8fdc
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
CSeq: 102 NOTIFY
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:21] --- (11 headers 0 lines) ---
[2011-11-03 06:47:21] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:21] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e2d5f16
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3568903f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 555690ff22901feb58d5ac8a193a199f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK461837fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0ac7f097
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0fc14b0f575ab173437a6b9c7191bb4d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5511b5e9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25eadd12
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49388bdc5648a1b71f8355b15dba7e08@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:21]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4169@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK60c75d6eF82DE90B
From: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
To: <sip:4169@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:21] --- (15 headers 0 lines) ---
[2011-11-03 06:47:21] Creating new subscription
[2011-11-03 06:47:21] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:21] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:21] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:21]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK60c75d6eF82DE90B;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
To: <sip:4169@customer.com>;tag=as34506259
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="051cc8a5"
Content-Length: 0


<------------>
[2011-11-03 06:47:21] Scheduling destruction of SIP dialog '330f082c-b8c37221-ae9a447a@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:21]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4169@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK914f9007B0ED29D8
From: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
To: <sip:4169@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="051cc8a5", uri="sip:4169@customer.com:5060", response="a30d83022bc7898eb05651c3e8b4ba5d", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:21] --- (16 headers 0 lines) ---
[2011-11-03 06:47:21] Creating new subscription
[2011-11-03 06:47:21] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:21] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:21] Looking for 4169 in default-local (domain customer.com:5060)
[2011-11-03 06:47:21] Scheduling destruction of SIP dialog '330f082c-b8c37221-ae9a447a@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:21]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK914f9007B0ED29D8;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
To: <sip:4169@customer.com>;tag=as34506259
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4169@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:21] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:21] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:21] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ca1ca29
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:21]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ca1ca29
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
CSeq: 102 NOTIFY
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:21] --- (11 headers 0 lines) ---
[2011-11-03 06:47:21] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:21] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK212afd08
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b7e9322
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7973f4bf5410cb832091ce2236afd267@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00824b79
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as33c2dadb
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4578625c0a78de0755e2371227044ead@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Really destroying SIP dialog '4578625c0a78de0755e2371227044ead@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:22] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:22] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK640fd1d2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as751be278
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5563252a4c04f84d096e42394aeecc7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.173:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKf41db1cbB7C73E42
From: "4118" <sip:4118@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4118@customer.com>
CSeq: 1 REGISTER
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (12 headers 0 lines) ---
[2011-11-03 06:47:22] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:22]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKf41db1cbB7C73E42;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4118@customer.com>;tag=as243663af
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.173
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23c8fdb7"
Content-Length: 0


<------------>
[2011-11-03 06:47:22] Scheduling destruction of SIP dialog 'c60da669-7b8a5940-3e53f5c7@172.30.0.173' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:22] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK177aa369
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5afc6a59
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 660db88c714ec9e32e173e9029ccb8e0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Really destroying SIP dialog '660db88c714ec9e32e173e9029ccb8e0@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:22] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK17ad3e46
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as244ea810
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28382b380911cff206cf607d79d861f3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.173:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKc6254afeAF3457E5
From: "4118" <sip:4118@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4118@customer.com>
CSeq: 2 REGISTER
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4118", realm="asterisk", nonce="23c8fdb7", uri="sip:customer.com:5060", response="dc1282fa4bb9d7c0044cd65111a66ace", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (13 headers 0 lines) ---
[2011-11-03 06:47:22] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:22]     -- Registered SIP '4118' at 172.30.0.173:5060
[2011-11-03 06:47:22] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
OPTIONS sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f25938a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as081791a9
To: <sip:4118@172.30.0.173:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a7d9fd4518b36fe0038adbd4e3e56c0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKc6254afeAF3457E5;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4118@customer.com>;tag=as243663af
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.173
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4118@172.30.0.173:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:22 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:22] Scheduling destruction of SIP dialog 'c60da669-7b8a5940-3e53f5c7@172.30.0.173' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:22] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f13ea27
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2eaf1dbe
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2b9cf7385f382b0b1f028d11307ecc5a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4113 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4137 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4138 (queued)
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:22] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:47:22] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:47:22] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK29b1c9bf
Max-Forwards: 70
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
Contact: <sip:4118@172.30.0.10:5060>
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4118">
<address uri="sip:4118@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4136
[2011-11-03 06:47:22]   == Extension Changed 4118[default-local] new state Idle for Notify User 4171 (queued)
[2011-11-03 06:47:22] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26dd51cd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a351821
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6bac251d1436dcf34a24c8c473951615@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Really destroying SIP dialog '6bac251d1436dcf34a24c8c473951615@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK15bf27f53AD9B61A
From: "4145" <sip:4145@customer.com>;tag=2D4C3755-7410F1FA
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 77724e5b-ba7e4288-79ca5031@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (15 headers 0 lines) ---
[2011-11-03 06:47:22] Creating new subscription
[2011-11-03 06:47:22] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:22] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:22] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:22]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK15bf27f53AD9B61A;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=2D4C3755-7410F1FA
To: <sip:4109@customer.com>;tag=as749d1041
Call-ID: 77724e5b-ba7e4288-79ca5031@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63b113fd"
Content-Length: 0


<------------>
[2011-11-03 06:47:22] Scheduling destruction of SIP dialog '77724e5b-ba7e4288-79ca5031@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:22] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d3f4e82
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10395db6
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71c4ba862d920c7d1d92ae0d01fbec3d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK62e38d7d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as703e8551
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c72da2679bf042c4bf491360d01b226@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Really destroying SIP dialog '5c72da2679bf042c4bf491360d01b226@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3f25938a
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as081791a9
To: "4118" <sip:4118@172.30.0.173:5060>;tag=468AA609-3B442AE0
CSeq: 102 OPTIONS
Call-ID: 3a7d9fd4518b36fe0038adbd4e3e56c0@172.30.0.10:5060
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (14 headers 0 lines) ---
[2011-11-03 06:47:22] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4118' is now Reachable. (167ms / 2000ms)
[2011-11-03 06:47:22] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK425cf67f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08e41776
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 68b9769f79fa6bef6cb884ca32af0104@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f0fd860
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as187410ad
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0e44634861a72fd41644f8cc292e9e97@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Really destroying SIP dialog '3a7d9fd4518b36fe0038adbd4e3e56c0@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK29b1c9bf
From: <sip:4118@customer.com>;tag=as384e4169
To: "4136" <sip:4136@customer.com>;tag=7F503DF6-11918A4F
CSeq: 105 NOTIFY
Call-ID: f94def59-2e17b9f2-9d7a134b@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (11 headers 0 lines) ---
[2011-11-03 06:47:22] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:22] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e2d5f16
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3568903f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 555690ff22901feb58d5ac8a193a199f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKf2d1f5a6851922F7
From: "4145" <sip:4145@customer.com>;tag=2D4C3755-7410F1FA
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 77724e5b-ba7e4288-79ca5031@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="63b113fd", uri="sip:4109@customer.com:5060", response="ff7d79eacbc2749025f0211ac7ac41b6", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (16 headers 0 lines) ---
[2011-11-03 06:47:22] Creating new subscription
[2011-11-03 06:47:22] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:22] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:22] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:22] Scheduling destruction of SIP dialog '77724e5b-ba7e4288-79ca5031@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:22]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKf2d1f5a6851922F7;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=2D4C3755-7410F1FA
To: <sip:4109@customer.com>;tag=as749d1041
Call-ID: 77724e5b-ba7e4288-79ca5031@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:22] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:22] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:22] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK177286e2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as749d1041
To: "4145" <sip:4145@customer.com>;tag=2D4C3755-7410F1FA
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 77724e5b-ba7e4288-79ca5031@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK177286e2
From: <sip:4109@customer.com>;tag=as749d1041
To: "4145" <sip:4145@customer.com>;tag=2D4C3755-7410F1FA
CSeq: 102 NOTIFY
Call-ID: 77724e5b-ba7e4288-79ca5031@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (11 headers 0 lines) ---
[2011-11-03 06:47:22] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:22] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK461837fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0ac7f097
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0fc14b0f575ab173437a6b9c7191bb4d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c1ff245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as511d16c5
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5511b5e9
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as25eadd12
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49388bdc5648a1b71f8355b15dba7e08@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK6144103d8E9EF3C6
From: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (15 headers 0 lines) ---
[2011-11-03 06:47:22] Creating new subscription
[2011-11-03 06:47:22] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:22] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:22] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:22]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK6144103d8E9EF3C6;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
To: <sip:4148@customer.com>;tag=as7a9dfdc2
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4981b755"
Content-Length: 0


<------------>
[2011-11-03 06:47:22] Scheduling destruction of SIP dialog '3a3db443-9199e1c4-78b99199@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:22] Really destroying SIP dialog '49388bdc5648a1b71f8355b15dba7e08@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK48d417525DF771BF
From: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="4981b755", uri="sip:4148@customer.com:5060", response="5c6a8a5a7805629d066fc59ffe584144", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (16 headers 0 lines) ---
[2011-11-03 06:47:22] Creating new subscription
[2011-11-03 06:47:22] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:22] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:22] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:22] Scheduling destruction of SIP dialog '3a3db443-9199e1c4-78b99199@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:22]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK48d417525DF771BF;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
To: <sip:4148@customer.com>;tag=as7a9dfdc2
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:22] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:22] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:22] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54917504
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:22] Reliably Transmitting (no NAT) to 172.30.0.198:5060:
OPTIONS sip:4142@172.30.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45acd86f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6b49d6b6
To: <sip:4142@172.30.0.198:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 74ece4da6a003fce2d4e971d4dd8eb4e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:22]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54917504
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
CSeq: 102 NOTIFY
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:22] --- (11 headers 0 lines) ---
[2011-11-03 06:47:22] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:22] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK212afd08
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b7e9322
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7973f4bf5410cb832091ce2236afd267@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45acd86f
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6b49d6b6
To: "4142" <sip:4142@172.30.0.198:5060>;tag=22E2A5D5-BFB25874
CSeq: 102 OPTIONS
Call-ID: 74ece4da6a003fce2d4e971d4dd8eb4e@172.30.0.10:5060
Contact: <sip:4142@172.30.0.198:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (14 headers 0 lines) ---
[2011-11-03 06:47:23] Really destroying SIP dialog '74ece4da6a003fce2d4e971d4dd8eb4e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.185:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.185:5060;branch=z9hG4bKf41db1cbB7C73E42
From: "4112" <sip:4112@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4112@customer.com>
CSeq: 1 REGISTER
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.185
Contact: <sip:4112@172.30.0.185:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (12 headers 0 lines) ---
[2011-11-03 06:47:23] Sending to 172.30.0.185:5060 (no NAT)
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.185:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.185:5060;branch=z9hG4bKf41db1cbB7C73E42;received=172.30.0.185
From: "4112" <sip:4112@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4112@customer.com>;tag=as6251a905
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.185
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6958f900"
Content-Length: 0


<------------>
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog 'c60da669-7b8a5940-3e53f5c7@172.30.0.185' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.185:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.185:5060;branch=z9hG4bKc6254afeAF3457E5
From: "4112" <sip:4112@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4112@customer.com>
CSeq: 2 REGISTER
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.185
Contact: <sip:4112@172.30.0.185:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4112", realm="asterisk", nonce="6958f900", uri="sip:customer.com:5060", response="839c09d73034c952bf99d3406f9c4046", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (13 headers 0 lines) ---
[2011-11-03 06:47:23] Sending to 172.30.0.185:5060 (no NAT)
[2011-11-03 06:47:23]     -- Registered SIP '4112' at 172.30.0.185:5060
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.185:5060:
OPTIONS sip:4112@172.30.0.185:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK643c517f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as12e41036
To: <sip:4112@172.30.0.185:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 03c210a6199662035c0870ca11f32b0a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.185:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.185:5060;branch=z9hG4bKc6254afeAF3457E5;received=172.30.0.185
From: "4112" <sip:4112@customer.com>;tag=1B935FED-BB7DC204
To: <sip:4112@customer.com>;tag=as6251a905
Call-ID: c60da669-7b8a5940-3e53f5c7@172.30.0.185
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4112@172.30.0.185:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:23 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog 'c60da669-7b8a5940-3e53f5c7@172.30.0.185' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:23] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK640fd1d2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as751be278
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5563252a4c04f84d096e42394aeecc7f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23] Really destroying SIP dialog '5563252a4c04f84d096e42394aeecc7f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:23] Retransmitting #6 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c88042b
Max-Forwards: 70
From: <sip:4130@customer.com>;tag=as410700c0
To: "4125" <sip:4125@customer.com>;tag=A4180A65-7047D79A
Contact: <sip:4130@172.30.0.10:5060>
Call-ID: e9f68b50-661d74e9-1f7900ce@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4130">
<address uri="sip:4130@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKefff39673D58A09E
From: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (15 headers 0 lines) ---
[2011-11-03 06:47:23] Creating new subscription
[2011-11-03 06:47:23] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:23] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:23] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKefff39673D58A09E;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
To: <sip:4148@customer.com>;tag=as551a6ae9
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0268c866"
Content-Length: 0


<------------>
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog 'b2085f85-8168a51c-caacd063@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:23] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:23] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK17ad3e46
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as244ea810
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28382b380911cff206cf607d79d861f3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e9f68b50-661d74e9-1f7900ce@172.30.0.168 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6458ms with no response
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74e673e6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4034d0af
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 02160f46356cd938791929bd38e806fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f13ea27
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2eaf1dbe
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2b9cf7385f382b0b1f028d11307ecc5a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK43181d91
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41af5370
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 77fcbb8012d7139161b070ef4aa1380a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.185:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK643c517f
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as12e41036
To: "4112" <sip:4112@172.30.0.185:5060>;tag=302584C3-720A92BA
CSeq: 102 OPTIONS
Call-ID: 03c210a6199662035c0870ca11f32b0a@172.30.0.10:5060
Contact: <sip:4112@172.30.0.185:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (14 headers 0 lines) ---
[2011-11-03 06:47:23] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4112' is now Reachable. (177ms / 2000ms)
[2011-11-03 06:47:23] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d3f4e82
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10395db6
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71c4ba862d920c7d1d92ae0d01fbec3d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23] Really destroying SIP dialog '03c210a6199662035c0870ca11f32b0a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK2e3516f46900F82D
From: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (15 headers 0 lines) ---
[2011-11-03 06:47:23] Creating new subscription
[2011-11-03 06:47:23] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:23] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:23] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK2e3516f46900F82D;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
To: <sip:4127@customer.com>;tag=as66a5e48b
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="00d77969"
Content-Length: 0


<------------>
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog '4f79f2-a0609d53-c7e55e20@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:23] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK425cf67f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08e41776
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 68b9769f79fa6bef6cb884ca32af0104@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f0fd860
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as187410ad
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0e44634861a72fd41644f8cc292e9e97@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK37c8705a23064401
From: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="0268c866", uri="sip:4148@customer.com:5060", response="6c922e6f7c020dbb6c92912587922c20", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (16 headers 0 lines) ---
[2011-11-03 06:47:23] Creating new subscription
[2011-11-03 06:47:23] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:23] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:23] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog 'b2085f85-8168a51c-caacd063@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK37c8705a23064401;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
To: <sip:4148@customer.com>;tag=as551a6ae9
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:23] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:23] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27b89375
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:23] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e2d5f16
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3568903f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 555690ff22901feb58d5ac8a193a199f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKa366dbe94AC74EFE
From: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="00d77969", uri="sip:4127@customer.com:5060", response="330ef9db5079feaca782c1d55984c8bf", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (16 headers 0 lines) ---
[2011-11-03 06:47:23] Creating new subscription
[2011-11-03 06:47:23] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:23] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:23] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog '4f79f2-a0609d53-c7e55e20@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKa366dbe94AC74EFE;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
To: <sip:4127@customer.com>;tag=as66a5e48b
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:23] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:23] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a7dbabf
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:23] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK461837fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0ac7f097
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0fc14b0f575ab173437a6b9c7191bb4d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23] Really destroying SIP dialog '0fc14b0f575ab173437a6b9c7191bb4d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27b89375
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
CSeq: 102 NOTIFY
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:23] --- (11 headers 0 lines) ---
[2011-11-03 06:47:23] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a7dbabf
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
CSeq: 102 NOTIFY
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (11 headers 0 lines) ---
[2011-11-03 06:47:23] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b9fea78
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e51ad2c
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6dd65d917bfe3c5468176e746f8e6791@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.180:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa97f0e1b9B822D24
From: "Office 106" <sip:4120@customer.com>;tag=3CAAD889-A0B35C92
To: <sip:4120@customer.com>
CSeq: 1 REGISTER
Call-ID: f190f9ad-ecdfb3b6-25029b3f@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (12 headers 0 lines) ---
[2011-11-03 06:47:23] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa97f0e1b9B822D24;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=3CAAD889-A0B35C92
To: <sip:4120@customer.com>;tag=as5decfed9
Call-ID: f190f9ad-ecdfb3b6-25029b3f@172.30.0.180
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6cf40f5c"
Content-Length: 0


<------------>
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog 'f190f9ad-ecdfb3b6-25029b3f@172.30.0.180' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:23] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c1ff245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as511d16c5
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.180:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKad6c67f29BB5297B
From: "Office 106" <sip:4120@customer.com>;tag=3CAAD889-A0B35C92
To: <sip:4120@customer.com>
CSeq: 2 REGISTER
Call-ID: f190f9ad-ecdfb3b6-25029b3f@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4120", realm="asterisk", nonce="6cf40f5c", uri="sip:customer.com:5060", response="7353fd73125e554b0a139db9c336303b", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (13 headers 0 lines) ---
[2011-11-03 06:47:23] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
OPTIONS sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fe0fb3e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2619adbc
To: <sip:4120@172.30.0.180:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6725bd2a7e040c2757a914e9156b2b10@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKad6c67f29BB5297B;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=3CAAD889-A0B35C92
To: <sip:4120@customer.com>;tag=as5decfed9
Call-ID: f190f9ad-ecdfb3b6-25029b3f@172.30.0.180
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4120@172.30.0.180:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:23 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog 'f190f9ad-ecdfb3b6-25029b3f@172.30.0.180' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59c3d7e4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1adc3c2b
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6e2009487cb5947900998f6d157559ff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKbff97bf0E13235
From: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (15 headers 0 lines) ---
[2011-11-03 06:47:23] Creating new subscription
[2011-11-03 06:47:23] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:23] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:23] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:23]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKbff97bf0E13235;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
To: <sip:4109@customer.com>;tag=as55cf4565
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a267256"
Content-Length: 0


<------------>
[2011-11-03 06:47:23] Scheduling destruction of SIP dialog '7fbe9b1e-930f247b-3fd4845c@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:23] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
OPTIONS sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK40437752
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1ec3afc8
To: <sip:4136@172.30.0.172:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4a8202c7271c6d2e7e554fd21078735f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:23]
<--- SIP read from UDP:172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3fe0fb3e
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2619adbc
To: "Office 106" <sip:4120@172.30.0.180:5060>;tag=EB89350D-D11CFF16
CSeq: 102 OPTIONS
Call-ID: 6725bd2a7e040c2757a914e9156b2b10@172.30.0.10:5060
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:23] --- (14 headers 0 lines) ---
[2011-11-03 06:47:23] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4120' is now Reachable. (153ms / 2000ms)
[2011-11-03 06:47:23] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK212afd08
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b7e9322
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7973f4bf5410cb832091ce2236afd267@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24] Really destroying SIP dialog '7973f4bf5410cb832091ce2236afd267@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4131 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4116 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4125 (queued)
[2011-11-03 06:47:24]   == Extension Changed 4120[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:47:24] Really destroying SIP dialog '6725bd2a7e040c2757a914e9156b2b10@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb4416e119B5AB2A
From: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="5a267256", uri="sip:4109@customer.com:5060", response="5705fdac6586d6ca8da89fcbb7e512e8", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (16 headers 0 lines) ---
[2011-11-03 06:47:24] Creating new subscription
[2011-11-03 06:47:24] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:24] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:24] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog '7fbe9b1e-930f247b-3fd4845c@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb4416e119B5AB2A;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
To: <sip:4109@customer.com>;tag=as55cf4565
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:24] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:24] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:24] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e457c2c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as55cf4565
To: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK40437752
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1ec3afc8
To: "4136" <sip:4136@172.30.0.172:5060>;tag=FDB207BD-A0D88D56
CSeq: 102 OPTIONS
Call-ID: 4a8202c7271c6d2e7e554fd21078735f@172.30.0.10:5060
Contact: <sip:4136@172.30.0.172:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (14 headers 0 lines) ---
[2011-11-03 06:47:24] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK17ad3e46
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as244ea810
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28382b380911cff206cf607d79d861f3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f13ea27
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2eaf1dbe
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2b9cf7385f382b0b1f028d11307ecc5a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24] Really destroying SIP dialog '4a8202c7271c6d2e7e554fd21078735f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa24f3a58721725F
From: "4118" <sip:4118@customer.com>;tag=346058B8-CD541EBF
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 92373b16-49de137d-11fdaa94@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (15 headers 0 lines) ---
[2011-11-03 06:47:24] Creating new subscription
[2011-11-03 06:47:24] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:24] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:24] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa24f3a58721725F;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=346058B8-CD541EBF
To: <sip:4109@customer.com>;tag=as5b9eef34
Call-ID: 92373b16-49de137d-11fdaa94@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c6166e4"
Content-Length: 0


<------------>
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog '92373b16-49de137d-11fdaa94@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:24] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e457c2c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as55cf4565
To: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:24] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74e673e6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4034d0af
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 02160f46356cd938791929bd38e806fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e457c2c
From: <sip:4109@customer.com>;tag=as55cf4565
To: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
CSeq: 102 NOTIFY
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (11 headers 0 lines) ---
[2011-11-03 06:47:24] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:24] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK43181d91
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41af5370
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 77fcbb8012d7139161b070ef4aa1380a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d3f4e82
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10395db6
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71c4ba862d920c7d1d92ae0d01fbec3d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK64a4d96f1FD5040C
From: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (15 headers 0 lines) ---
[2011-11-03 06:47:24] Creating new subscription
[2011-11-03 06:47:24] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:24] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:24] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK64a4d96f1FD5040C;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
To: <sip:4122@customer.com>;tag=as1656283f
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f186ac7"
Content-Length: 0


<------------>
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog '95b77765-dfcb40ca-f5dfb34b@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:24] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK425cf67f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08e41776
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 68b9769f79fa6bef6cb884ca32af0104@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f0fd860
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as187410ad
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0e44634861a72fd41644f8cc292e9e97@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK54cedf5bBB42C0D2
From: "4118" <sip:4118@customer.com>;tag=346058B8-CD541EBF
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 92373b16-49de137d-11fdaa94@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="2c6166e4", uri="sip:4109@customer.com:5060", response="4586791e25ebb000c0cf8caf70d3f660", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (16 headers 0 lines) ---
[2011-11-03 06:47:24] Creating new subscription
[2011-11-03 06:47:24] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:24] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:24] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog '92373b16-49de137d-11fdaa94@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK54cedf5bBB42C0D2;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=346058B8-CD541EBF
To: <sip:4109@customer.com>;tag=as5b9eef34
Call-ID: 92373b16-49de137d-11fdaa94@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:24] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:24] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:24] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d08d9b7
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as5b9eef34
To: "4118" <sip:4118@customer.com>;tag=346058B8-CD541EBF
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 92373b16-49de137d-11fdaa94@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:24] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e2d5f16
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3568903f
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 555690ff22901feb58d5ac8a193a199f@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24] Really destroying SIP dialog '555690ff22901feb58d5ac8a193a199f@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5e457c2c
From: <sip:4109@customer.com>;tag=as55cf4565
To: "Office 131" <sip:4113@customer.com>;tag=C24109A6-70F477A3
CSeq: 102 NOTIFY
Call-ID: 7fbe9b1e-930f247b-3fd4845c@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (11 headers 0 lines) ---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK8854b4b828A206A1
From: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="5f186ac7", uri="sip:4122@customer.com:5060", response="01077d837bfb8e0ddc4b30121bb609bf", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (16 headers 0 lines) ---
[2011-11-03 06:47:24] Creating new subscription
[2011-11-03 06:47:24] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:24] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:24] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog '95b77765-dfcb40ca-f5dfb34b@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK8854b4b828A206A1;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
To: <sip:4122@customer.com>;tag=as1656283f
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:24] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:24] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:24] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f32a0e4
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as1656283f
To: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d08d9b7
From: <sip:4109@customer.com>;tag=as5b9eef34
To: "4118" <sip:4118@customer.com>;tag=346058B8-CD541EBF
CSeq: 102 NOTIFY
Call-ID: 92373b16-49de137d-11fdaa94@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:24] --- (11 headers 0 lines) ---
[2011-11-03 06:47:24] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.191:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKfd37520bE73F397A
From: "empty station" <sip:4119@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4119@customer.com>
CSeq: 1 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (12 headers 0 lines) ---
[2011-11-03 06:47:24] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKfd37520bE73F397A;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4119@customer.com>;tag=as0242f418
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.191
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31c94214"
Content-Length: 0


<------------>
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.191' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:24] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b9fea78
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e51ad2c
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6dd65d917bfe3c5468176e746f8e6791@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKbd1cf69f917D5BA8
From: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (15 headers 0 lines) ---
[2011-11-03 06:47:24] Creating new subscription
[2011-11-03 06:47:24] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:24] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:24] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKbd1cf69f917D5BA8;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
To: <sip:4148@customer.com>;tag=as5371e3da
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26a91d8d"
Content-Length: 0


<------------>
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog 'a6926e31-5b106e3a-a40b9bc3@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:24] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f32a0e4
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as1656283f
To: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:24] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c1ff245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as511d16c5
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f32a0e4
From: <sip:4122@customer.com>;tag=as1656283f
To: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
CSeq: 102 NOTIFY
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (11 headers 0 lines) ---
[2011-11-03 06:47:24] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.191:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK3501afc4580F5E2B
From: "empty station" <sip:4119@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4119@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4119", realm="asterisk", nonce="31c94214", uri="sip:customer.com:5060", response="be23b58d98caea51c9f2e3dfb7b81445", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (13 headers 0 lines) ---
[2011-11-03 06:47:24] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:24] NOTICE[9661]: chan_sip.c:25638 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2011-11-03 06:47:24] Really destroying SIP dialog '0e44634861a72fd41644f8cc292e9e97@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:24] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45a03c42
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69d3691d
To: <sip:4119@172.30.0.191:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78f97de535512a91210b0d241064e799@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK3501afc4580F5E2B;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4119@customer.com>;tag=as0242f418
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.191
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4119@172.30.0.191:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:24 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.191' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:24] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59c3d7e4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1adc3c2b
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6e2009487cb5947900998f6d157559ff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:24]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK9cf836cc17AA7F55
From: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="26a91d8d", uri="sip:4148@customer.com:5060", response="30a7e634bd3136148db37fb5f0179fbe", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:24] --- (16 headers 0 lines) ---
[2011-11-03 06:47:24] Creating new subscription
[2011-11-03 06:47:24] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:24] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:24] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:24] Scheduling destruction of SIP dialog 'a6926e31-5b106e3a-a40b9bc3@172.30.0.180' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK9cf836cc17AA7F55;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
To: <sip:4148@customer.com>;tag=as5371e3da
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:25] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:47:25] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:47:25] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5c3d7184
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0f32a0e4
From: <sip:4122@customer.com>;tag=as1656283f
To: "4145" <sip:4145@customer.com>;tag=919ED757-B49D13D4
CSeq: 102 NOTIFY
Call-ID: 95b77765-dfcb40ca-f5dfb34b@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (11 headers 0 lines) ---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb1e668773FC3C708
From: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (15 headers 0 lines) ---
[2011-11-03 06:47:25] Creating new subscription
[2011-11-03 06:47:25] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:25] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:25] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKb1e668773FC3C708;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
To: <sip:4127@customer.com>;tag=as4fa3ce12
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="481084ce"
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog '6986012d-1e6eb376-e42a19b3@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.164:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKfd37520bE73F397A
From: "4124" <sip:4124@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4124@customer.com>
CSeq: 1 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (12 headers 0 lines) ---
[2011-11-03 06:47:25] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKfd37520bE73F397A;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4124@customer.com>;tag=as056000e4
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.164
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="123b2a0e"
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.164' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:25] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK17ad3e46
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as244ea810
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28382b380911cff206cf607d79d861f3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5c3d7184
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25] Really destroying SIP dialog '28382b380911cff206cf607d79d861f3@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45a03c42
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69d3691d
To: "empty station" <sip:4119@172.30.0.191:5060>;tag=B769A0D1-C8120290
CSeq: 102 OPTIONS
Call-ID: 78f97de535512a91210b0d241064e799@172.30.0.10:5060
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (14 headers 0 lines) ---
[2011-11-03 06:47:25] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4119' is now Reachable. (298ms / 2000ms)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4113 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:25] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:47:25] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:47:25] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4173
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4131 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4171 (queued)
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:25] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:25] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:25] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3eba0bef
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25]   == Extension Changed 4119[default-local] new state Idle for Notify User 4113
[2011-11-03 06:47:25] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f13ea27
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2eaf1dbe
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2b9cf7385f382b0b1f028d11307ecc5a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25] Really destroying SIP dialog '2b9cf7385f382b0b1f028d11307ecc5a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:25] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25] Retransmitting #6 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
Max-Forwards: 70
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
Contact: <sip:4113@172.30.0.10:5060>
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4113">
<address uri="sip:4113@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74e673e6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4034d0af
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 02160f46356cd938791929bd38e806fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25] Really destroying SIP dialog '78f97de535512a91210b0d241064e799@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.179:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKfd37520bE73F397A
From: "Leon Zhong" <sip:4134@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4134@customer.com>
CSeq: 1 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (12 headers 0 lines) ---
[2011-11-03 06:47:25] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKfd37520bE73F397A;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4134@customer.com>;tag=as2e70d8d2
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.179
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="186215a1"
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.179' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:25] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK43181d91
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41af5370
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 77fcbb8012d7139161b070ef4aa1380a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d3f4e82
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10395db6
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71c4ba862d920c7d1d92ae0d01fbec3d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25] Really destroying SIP dialog '71c4ba862d920c7d1d92ae0d01fbec3d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5c3d7184
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
CSeq: 102 NOTIFY
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (11 headers 0 lines) ---
[2011-11-03 06:47:25] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:25] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 1f0d3c30-1aa6f385-50263dbe@172.30.0.181 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6423ms with no response
[2011-11-03 06:47:25] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3eba0bef
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK425cf67f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as08e41776
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 68b9769f79fa6bef6cb884ca32af0104@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25] Really destroying SIP dialog '68b9769f79fa6bef6cb884ca32af0104@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKed5c4ff4F61BE789
From: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="481084ce", uri="sip:4127@customer.com:5060", response="4ce01d43395ded64e3c574d7fe21ff67", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (16 headers 0 lines) ---
[2011-11-03 06:47:25] Creating new subscription
[2011-11-03 06:47:25] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:25] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:25] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog '6986012d-1e6eb376-e42a19b3@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKed5c4ff4F61BE789;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
To: <sip:4127@customer.com>;tag=as4fa3ce12
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:25] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:25] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:25] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa4198df938A9B5D0
From: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (15 headers 0 lines) ---
[2011-11-03 06:47:25] Creating new subscription
[2011-11-03 06:47:25] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:25] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:25] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa4198df938A9B5D0;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
To: <sip:4127@customer.com>;tag=as4fe58b4c
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5bf2069a"
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog '2e56d757-73b8c18e-38727375@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:25] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3eba0bef
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.164:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK3501afc4580F5E2B
From: "4124" <sip:4124@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4124@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4124", realm="asterisk", nonce="123b2a0e", uri="sip:customer.com:5060", response="6e3318bb7c5151c7af93a02a05dc19f9", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (13 headers 0 lines) ---
[2011-11-03 06:47:25] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:25] NOTICE[9661]: chan_sip.c:25638 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2011-11-03 06:47:25] Really destroying SIP dialog '77fcbb8012d7139161b070ef4aa1380a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:25] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
OPTIONS sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK30901e8c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f0e78a7
To: <sip:4124@172.30.0.164:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a61a94e78d78897144e47ef4076b88d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK3501afc4580F5E2B;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4124@customer.com>;tag=as056000e4
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.164
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4124@172.30.0.164:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:25 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.164' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:25] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b9fea78
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e51ad2c
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6dd65d917bfe3c5468176e746f8e6791@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5c3d7184
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
CSeq: 102 NOTIFY
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (11 headers 0 lines) ---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.199:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKfd37520bE73F397A
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>
CSeq: 1 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (12 headers 0 lines) ---
[2011-11-03 06:47:25] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKfd37520bE73F397A;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>;tag=as519f39cd
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3351d64e"
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.199' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:25] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:25] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c1ff245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as511d16c5
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.171:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK6f359e4b89F18D22
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>
CSeq: 1 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (12 headers 0 lines) ---
[2011-11-03 06:47:25] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK6f359e4b89F18D22;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>;tag=as0c17639a
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f3c9f72"
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.171' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3eba0bef
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
CSeq: 103 NOTIFY
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (11 headers 0 lines) ---
[2011-11-03 06:47:25] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:25] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59c3d7e4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1adc3c2b
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6e2009487cb5947900998f6d157559ff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.181:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKfd37520bE73F397A
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>
CSeq: 1 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (12 headers 0 lines) ---
[2011-11-03 06:47:25] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:25]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKfd37520bE73F397A;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>;tag=as27ac367e
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1089ebd7"
Content-Length: 0


<------------>
[2011-11-03 06:47:25] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.181' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:25]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d4d3b62
From: <sip:4113@customer.com>;tag=as779a1bc4
To: "Office 107" <sip:4141@customer.com>;tag=32EB6011-1806BC3A
CSeq: 105 NOTIFY
Call-ID: 1f0d3c30-1aa6f385-50263dbe@172.30.0.181
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:25] --- (10 headers 0 lines) ---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.179:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK3501afc4580F5E2B
From: "Leon Zhong" <sip:4134@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4134@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4134", realm="asterisk", nonce="186215a1", uri="sip:customer.com:5060", response="0eb1705adc8b0182991976eaef3f14be", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (13 headers 0 lines) ---
[2011-11-03 06:47:26] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:26] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ce84aec
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a039ede
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78db473a0bc746d26a16c7661577d0ee@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:26]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK3501afc4580F5E2B;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4134@customer.com>;tag=as2e70d8d2
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.179
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4134@172.30.0.179:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:26 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:26] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.179' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK43181d91
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as41af5370
To: "4124" <sip:4124@172.30.0.164:5060>;tag=B769A0D1-C8120290
CSeq: 102 OPTIONS
Call-ID: 77fcbb8012d7139161b070ef4aa1380a@172.30.0.10:5060
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (14 headers 0 lines) ---
[2011-11-03 06:47:26] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3eba0bef
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
CSeq: 103 NOTIFY
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (11 headers 0 lines) ---
[2011-11-03 06:47:26] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
CSeq: 102 NOTIFY
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (11 headers 0 lines) ---
[2011-11-03 06:47:26] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK69fb6b975DFD67A6
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (15 headers 0 lines) ---
[2011-11-03 06:47:26] Creating new subscription
[2011-11-03 06:47:26] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:26] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:47:26] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:26]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK69fb6b975DFD67A6;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as2406f454
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="359ff62b"
Content-Length: 0


<------------>
[2011-11-03 06:47:26] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:26] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74e673e6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4034d0af
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 02160f46356cd938791929bd38e806fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKeeb91c0c158B1A53
From: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="5bf2069a", uri="sip:4127@customer.com:5060", response="09b4c391c72bf834b12121d8d477444f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (16 headers 0 lines) ---
[2011-11-03 06:47:26] Creating new subscription
[2011-11-03 06:47:26] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:26] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:26] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:26] Scheduling destruction of SIP dialog '2e56d757-73b8c18e-38727375@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:26]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKeeb91c0c158B1A53;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
To: <sip:4127@customer.com>;tag=as4fe58b4c
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:26] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:26] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:26] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
CSeq: 102 NOTIFY
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (11 headers 0 lines) ---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3eba0bef
From: <sip:4119@customer.com>;tag=as68f65716
To: "Office 131" <sip:4113@customer.com>;tag=6438CE74-9573E809
CSeq: 103 NOTIFY
Call-ID: d62861ca-ab518b97-a964f0a8@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (11 headers 0 lines) ---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK30901e8c
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3f0e78a7
To: "4124" <sip:4124@172.30.0.164:5060>;tag=5DFD67A6-E5747D
CSeq: 102 OPTIONS
Call-ID: 7a61a94e78d78897144e47ef4076b88d@172.30.0.10:5060
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (14 headers 0 lines) ---
[2011-11-03 06:47:26] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4124' is now Reachable. (804ms / 2000ms)
[2011-11-03 06:47:26]   == Extension Changed 4124[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:26]   == Extension Changed 4124[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:26]   == Extension Changed 4124[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:26]   == Extension Changed 4124[default-local] new state Idle for Notify User 4135 (queued)
[2011-11-03 06:47:26]   == Extension Changed 4124[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:26]   == Extension Changed 4124[default-local] new state Idle for Notify User 4138 (queued)
[2011-11-03 06:47:26]   == Extension Changed 4124[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:26] Really destroying SIP dialog '7a61a94e78d78897144e47ef4076b88d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKcd95db565444C6E7
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (15 headers 0 lines) ---
[2011-11-03 06:47:26] Creating new subscription
[2011-11-03 06:47:26] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:26] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:26] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:26]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKcd95db565444C6E7;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>;tag=as2d5324ae
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04900da2"
Content-Length: 0


<------------>
[2011-11-03 06:47:26] Scheduling destruction of SIP dialog '752bdc24-7fc5859d-f2566aa2@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:26]     -- Refreshing DNS lookups.
[2011-11-03 06:47:26] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:26] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b9fea78
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e51ad2c
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6dd65d917bfe3c5468176e746f8e6791@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.199:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKfd37520bE73F397A
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>
CSeq: 1 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (12 headers 0 lines) ---
[2011-11-03 06:47:26] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:26]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKfd37520bE73F397A;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>;tag=as519f39cd
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3351d64e"
Content-Length: 0


<------------>
[2011-11-03 06:47:26] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.199' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.199:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK3501afc4580F5E2B
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4116", realm="asterisk", nonce="3351d64e", uri="sip:customer.com:5060", response="64cfcce5b66c6ba2e45c33f453be62f5", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (13 headers 0 lines) ---
[2011-11-03 06:47:26] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:26] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c4af326
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7f6f144d
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 173a569a0e8c7dd35564622379d63e16@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:26]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK3501afc4580F5E2B;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>;tag=as519f39cd
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4116@172.30.0.199:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:26 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:26] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.199' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:26] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c1ff245
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as511d16c5
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:26] Really destroying SIP dialog '60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
CSeq: 102 NOTIFY
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (11 headers 0 lines) ---
[2011-11-03 06:47:26] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:26] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59c3d7e4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1adc3c2b
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6e2009487cb5947900998f6d157559ff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c1ff245
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as511d16c5
To: "Office 107" <sip:4141@172.30.0.181:5060>;tag=580F5E2B-76F3C9A
CSeq: 102 OPTIONS
Call-ID: 60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (14 headers 0 lines) ---
[2011-11-03 06:47:26]
<--- SIP read from UDP:172.30.0.171:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK6f359e4b89F18D22
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>
CSeq: 1 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:26] --- (12 headers 0 lines) ---
[2011-11-03 06:47:26] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:26]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK6f359e4b89F18D22;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>;tag=as0c17639a
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f3c9f72"
Content-Length: 0


<------------>
[2011-11-03 06:47:26] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.171' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.179:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK3501afc4580F5E2B
From: "Leon Zhong" <sip:4134@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4134@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4134", realm="asterisk", nonce="186215a1", uri="sip:customer.com:5060", response="0eb1705adc8b0182991976eaef3f14be", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (13 headers 0 lines) ---
[2011-11-03 06:47:27] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:27] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Leon Zhong" <sip:4134@customer.com>;tag=422F5785-87B9D2A4'
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK3501afc4580F5E2B;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4134@customer.com>;tag=as2e70d8d2
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.179
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="279ef68f", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.179' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.171:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKfa669a087F11BAB
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>
CSeq: 2 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4106", realm="asterisk", nonce="2f3c9f72", uri="sip:customer.com:5060", response="0606e4bf020a258c7e3b06545f19bc05", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (13 headers 0 lines) ---
[2011-11-03 06:47:27] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:27] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ca9d9d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76eadd43
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b395e407902055246a6058a2663a1da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKfa669a087F11BAB;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>;tag=as0c17639a
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4106@172.30.0.171:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:27 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.171' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:27] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
OPTIONS sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ce84aec
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a039ede
To: <sip:4134@172.30.0.179:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78db473a0bc746d26a16c7661577d0ee@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:27] Retransmitting #4 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:27] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK74e673e6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4034d0af
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 02160f46356cd938791929bd38e806fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:27] Really destroying SIP dialog '02160f46356cd938791929bd38e806fe@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.181:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKfd37520bE73F397A
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>
CSeq: 1 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (12 headers 0 lines) ---
[2011-11-03 06:47:27] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKfd37520bE73F397A;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>;tag=as27ac367e
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1089ebd7"
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.181' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.170:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK6f359e4b89F18D22
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>
CSeq: 1 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (12 headers 0 lines) ---
[2011-11-03 06:47:27] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK6f359e4b89F18D22;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>;tag=as6f9f5b94
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a9d5d96"
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.170' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.181:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKb769a0d1C8120290
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4141", realm="asterisk", nonce="1089ebd7", uri="sip:customer.com:5060", response="cc93ea361f1cb306819e9318afac8f47", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (13 headers 0 lines) ---
[2011-11-03 06:47:27] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:27] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK66dd0c7e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1544bd08
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0b51238318aa5c995c74cb7a0f81b0a1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKb769a0d1C8120290;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>;tag=as27ac367e
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4141@172.30.0.181:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:27 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.181' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:27] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK69fb6b975DFD67A6
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (15 headers 0 lines) ---
[2011-11-03 06:47:27] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:27] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK69fb6b975DFD67A6;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as2406f454
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="359ff62b"
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ce84aec
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a039ede
To: "Leon Zhong" <sip:4134@172.30.0.179:5060>;tag=B769A0D1-C8120290
CSeq: 102 OPTIONS
Call-ID: 78db473a0bc746d26a16c7661577d0ee@172.30.0.10:5060
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (14 headers 0 lines) ---
[2011-11-03 06:47:27] NOTICE[9661]: chan_sip.c:20246 handle_response_peerpoke: Peer '4134' is now Reachable. (1546ms / 2000ms)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4128 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4113 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4138 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4135 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4171 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4116 (queued)
[2011-11-03 06:47:27] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:47:27] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:47:27] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:27]   == Extension Changed 4134[default-local] new state Idle for Notify User 4136
[2011-11-03 06:47:27] Really destroying SIP dialog '78db473a0bc746d26a16c7661577d0ee@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2c56b55eC99118E7
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (15 headers 0 lines) ---
[2011-11-03 06:47:27] Creating new subscription
[2011-11-03 06:47:27] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:27] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:27] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2c56b55eC99118E7;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>;tag=as4babbcd3
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6196ffa5"
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog '444de9f0-e7e16879-85ffd482@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:27] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b9fea78
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e51ad2c
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6dd65d917bfe3c5468176e746f8e6791@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:27] Really destroying SIP dialog '6dd65d917bfe3c5468176e746f8e6791@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKca96fd4a52D283F1
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (15 headers 0 lines) ---
[2011-11-03 06:47:27] Creating new subscription
[2011-11-03 06:47:27] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:27] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:27] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKca96fd4a52D283F1;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>;tag=as536f75cd
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03cec172"
Content-Length: 0


<------------>
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog 'f8c19d48-cf6d684f-190b3406@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:27] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:27] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c4af326
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7f6f144d
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 173a569a0e8c7dd35564622379d63e16@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:27] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59c3d7e4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1adc3c2b
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6e2009487cb5947900998f6d157559ff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:27] Really destroying SIP dialog '6e2009487cb5947900998f6d157559ff@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:27]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKeeb91c0c158B1A53
From: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="5bf2069a", uri="sip:4127@customer.com:5060", response="09b4c391c72bf834b12121d8d477444f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:27] --- (16 headers 0 lines) ---
[2011-11-03 06:47:27] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:27] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:27] Scheduling destruction of SIP dialog '2e56d757-73b8c18e-38727375@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:27]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKeeb91c0c158B1A53;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
To: <sip:4127@customer.com>;tag=as4fe58b4c
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:27] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:27] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:27] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:27] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK115356e4
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
CSeq: 102 NOTIFY
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (11 headers 0 lines) ---
[2011-11-03 06:47:28] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKcd95db565444C6E7
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (15 headers 0 lines) ---
[2011-11-03 06:47:28] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:28] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKcd95db565444C6E7;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>;tag=as2d5324ae
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04900da2"
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog '752bdc24-7fc5859d-f2566aa2@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.197:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK6f359e4b89F18D22
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>
CSeq: 1 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (12 headers 0 lines) ---
[2011-11-03 06:47:28] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK6f359e4b89F18D22;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>;tag=as4f450cb4
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="331f8cba"
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.197' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:28] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKba6e47326F5AB8A9
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="359ff62b", uri="sip:4148@customer.com:5060", response="fd80276399529baaf3d2545c8d1cadd0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (16 headers 0 lines) ---
[2011-11-03 06:47:28] Creating new subscription
[2011-11-03 06:47:28] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:28] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:28] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKba6e47326F5AB8A9;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as2406f454
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:28] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:28] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:28] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
CSeq: 102 NOTIFY
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:28] --- (11 headers 0 lines) ---
[2011-11-03 06:47:28] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:28] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ca9d9d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76eadd43
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b395e407902055246a6058a2663a1da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.199:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK3501afc4580F5E2B
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4116", realm="asterisk", nonce="3351d64e", uri="sip:customer.com:5060", response="64cfcce5b66c6ba2e45c33f453be62f5", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (13 headers 0 lines) ---
[2011-11-03 06:47:28] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:28] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4'
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK3501afc4580F5E2B;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4116@customer.com>;tag=as519f39cd
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.199
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60ddb1ce", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.199' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:28] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa10f1dc810B9183
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (15 headers 0 lines) ---
[2011-11-03 06:47:28] Creating new subscription
[2011-11-03 06:47:28] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:28] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:47:28] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa10f1dc810B9183;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as603feaf0
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63e41dbb"
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.171:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKfa669a087F11BAB
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>
CSeq: 2 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4106", realm="asterisk", nonce="2f3c9f72", uri="sip:customer.com:5060", response="0606e4bf020a258c7e3b06545f19bc05", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (13 headers 0 lines) ---
[2011-11-03 06:47:28] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:28] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0'
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKfa669a087F11BAB;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4106@customer.com>;tag=as0c17639a
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.171
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="754d678c", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.171' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:28] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.170:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK6f359e4b89F18D22
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>
CSeq: 1 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (12 headers 0 lines) ---
[2011-11-03 06:47:28] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK6f359e4b89F18D22;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>;tag=as6f9f5b94
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a9d5d96"
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.170' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:28] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK66dd0c7e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1544bd08
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0b51238318aa5c995c74cb7a0f81b0a1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:28] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.181:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKb769a0d1C8120290
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>
CSeq: 2 REGISTER
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4141", realm="asterisk", nonce="1089ebd7", uri="sip:customer.com:5060", response="cc93ea361f1cb306819e9318afac8f47", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (13 headers 0 lines) ---
[2011-11-03 06:47:28] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:28] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4'
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKb769a0d1C8120290;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=422F5785-87B9D2A4
To: <sip:4141@customer.com>;tag=as27ac367e
Call-ID: ab321eb1-229b970-7cbd5377@172.30.0.181
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="747d805d", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.181' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2c56b55eC99118E7
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (15 headers 0 lines) ---
[2011-11-03 06:47:28] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:28] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2c56b55eC99118E7;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>;tag=as4babbcd3
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6196ffa5"
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog '444de9f0-e7e16879-85ffd482@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKca96fd4a52D283F1
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (15 headers 0 lines) ---
[2011-11-03 06:47:28] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:28] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKca96fd4a52D283F1;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>;tag=as536f75cd
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03cec172"
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog 'f8c19d48-cf6d684f-190b3406@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
CSeq: 102 NOTIFY
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:28] --- (11 headers 0 lines) ---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK459d704385C4A650
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="04900da2", uri="sip:4108@customer.com:5060", response="1e9984aaf8dd8ef208ce17faf42e5488", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (16 headers 0 lines) ---
[2011-11-03 06:47:28] Creating new subscription
[2011-11-03 06:47:28] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:28] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:28] Looking for 4108 in default-local (domain customer.com:5060)
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog '752bdc24-7fc5859d-f2566aa2@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK459d704385C4A650;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>;tag=as2d5324ae
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4108@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:28] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:28] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:28] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK15ba396e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:28] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c4af326
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7f6f144d
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 173a569a0e8c7dd35564622379d63e16@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:28]
<--- SIP read from UDP:172.30.0.197:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK6f359e4b89F18D22
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>
CSeq: 1 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:28] --- (12 headers 0 lines) ---
[2011-11-03 06:47:28] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:28]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK6f359e4b89F18D22;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>;tag=as4f450cb4
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="331f8cba"
Content-Length: 0


<------------>
[2011-11-03 06:47:28] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.197' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:28] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK15ba396e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc4c9600298E542F
From: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (15 headers 0 lines) ---
[2011-11-03 06:47:29] Creating new subscription
[2011-11-03 06:47:29] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:29] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:29] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc4c9600298E542F;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
To: <sip:4122@customer.com>;tag=as5ad25dc4
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31be8868"
Content-Length: 0


<------------>
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog '979b6b20-c0d5d25-acdb9cce@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c4af326
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7f6f144d
To: "4116" <sip:4116@172.30.0.199:5060>;tag=B769A0D1-C8120290
CSeq: 102 OPTIONS
Call-ID: 173a569a0e8c7dd35564622379d63e16@172.30.0.10:5060
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (14 headers 0 lines) ---
[2011-11-03 06:47:29] Really destroying SIP dialog '173a569a0e8c7dd35564622379d63e16@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c1ff245
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as511d16c5
To: "Office 107" <sip:4141@172.30.0.181:5060>;tag=580F5E2B-76F3C9A
CSeq: 102 OPTIONS
Call-ID: 60feba6a4ee6f0d62aec8e012a9904ed@172.30.0.10:5060
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (14 headers 0 lines) ---
[2011-11-03 06:47:29] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK15ba396e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKba6e47326F5AB8A9
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="359ff62b", uri="sip:4148@customer.com:5060", response="fd80276399529baaf3d2545c8d1cadd0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (16 headers 0 lines) ---
[2011-11-03 06:47:29] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:29] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKba6e47326F5AB8A9;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as2406f454
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:29] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:29] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:29] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6394da8c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ca9d9d3
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76eadd43
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6b395e407902055246a6058a2663a1da@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.184:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bK8a59538f27A23744
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>
CSeq: 1 REGISTER
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (12 headers 0 lines) ---
[2011-11-03 06:47:29] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bK8a59538f27A23744;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>;tag=as1de8928a
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6942dad8"
Content-Length: 0


<------------>
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog '8d474b1d-d46049ea-53ee6bb@172.30.0.184' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa10f1dc810B9183
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (15 headers 0 lines) ---
[2011-11-03 06:47:29] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:29] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa10f1dc810B9183;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as603feaf0
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63e41dbb"
Content-Length: 0


<------------>
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
CSeq: 102 NOTIFY
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:29] --- (11 headers 0 lines) ---
[2011-11-03 06:47:29] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK66dd0c7e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1544bd08
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0b51238318aa5c995c74cb7a0f81b0a1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK69fb6b975DFD67A6
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (15 headers 0 lines) ---
[2011-11-03 06:47:29] Creating new subscription
[2011-11-03 06:47:29] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:29] list_route: hop: <sip:4134@172.30.0.179:5060>
[2011-11-03 06:47:29] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK69fb6b975DFD67A6;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as5e3b968c
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2df187d2"
Content-Length: 0


<------------>
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:29] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK15ba396e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK646cb059CC3CFAAE
From: "4145" <sip:4145@customer.com>;tag=86770D49-DA6504DE
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f064cb5f-1e17ff3c-d80a02d5@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (15 headers 0 lines) ---
[2011-11-03 06:47:29] Creating new subscription
[2011-11-03 06:47:29] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:29] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:29] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK646cb059CC3CFAAE;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=86770D49-DA6504DE
To: <sip:4115@customer.com>;tag=as2011c671
Call-ID: f064cb5f-1e17ff3c-d80a02d5@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f67bad0"
Content-Length: 0


<------------>
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog 'f064cb5f-1e17ff3c-d80a02d5@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:29] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6394da8c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK459d704385C4A650
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="04900da2", uri="sip:4108@customer.com:5060", response="1e9984aaf8dd8ef208ce17faf42e5488", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (16 headers 0 lines) ---
[2011-11-03 06:47:29] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:29] Looking for 4108 in default-local (domain customer.com:5060)
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog '752bdc24-7fc5859d-f2566aa2@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK459d704385C4A650;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
To: <sip:4108@customer.com>;tag=as2d5324ae
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4108@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:29] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:29] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:29] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK267c9de3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKa22f44021EF68055
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (15 headers 0 lines) ---
[2011-11-03 06:47:29] Creating new subscription
[2011-11-03 06:47:29] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:29] list_route: hop: <sip:4106@172.30.0.171:5060>
[2011-11-03 06:47:29] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKa22f44021EF68055;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2ff9da69
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a46154f"
Content-Length: 0


<------------>
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog '36b2a714-5b62726f-18b0f756@172.30.0.171' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:29]
<--- SIP read from UDP:172.30.0.175:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK487347976C342DFC
From: "4140" <sip:4140@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4140@customer.com>
CSeq: 1 REGISTER
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:29] --- (12 headers 0 lines) ---
[2011-11-03 06:47:29] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:47:29]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK487347976C342DFC;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4140@customer.com>;tag=as328acdbf
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.175
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37b3c6a1"
Content-Length: 0


<------------>
[2011-11-03 06:47:29] Scheduling destruction of SIP dialog 'e7b8eeb5-a95f4612-4fb66da3@172.30.0.175' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:29] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK267c9de3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2ca9d9d3
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76eadd43
To: "4106" <sip:4106@172.30.0.171:5060>;tag=F0B4E548-4C6094F3
CSeq: 102 OPTIONS
Call-ID: 6b395e407902055246a6058a2663a1da@172.30.0.10:5060
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (14 headers 0 lines) ---
[2011-11-03 06:47:30] Really destroying SIP dialog '6b395e407902055246a6058a2663a1da@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc4c9600298E542F
From: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (15 headers 0 lines) ---
[2011-11-03 06:47:30] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:30] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc4c9600298E542F;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
To: <sip:4122@customer.com>;tag=as5ad25dc4
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31be8868"
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog '979b6b20-c0d5d25-acdb9cce@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:30] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK267c9de3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ce84aec
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7a039ede
To: "Leon Zhong" <sip:4134@172.30.0.179:5060>;tag=B769A0D1-C8120290
CSeq: 102 OPTIONS
Call-ID: 78db473a0bc746d26a16c7661577d0ee@172.30.0.10:5060
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (14 headers 0 lines) ---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKfa669a087F11BAB
From: "Office 105" <sip:4131@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (15 headers 0 lines) ---
[2011-11-03 06:47:30] Creating new subscription
[2011-11-03 06:47:30] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:30] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:47:30] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKfa669a087F11BAB;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as6af9e4c3
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16cc0265"
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog 'a22f4402-1ef68055-36b2a714@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:30] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6394da8c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK69fb6b975DFD67A6
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (15 headers 0 lines) ---
[2011-11-03 06:47:30] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:30] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK69fb6b975DFD67A6;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as5e3b968c
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2df187d2"
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:30] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.184:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bK8a59538f27A23744
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>
CSeq: 1 REGISTER
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (12 headers 0 lines) ---
[2011-11-03 06:47:30] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bK8a59538f27A23744;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>;tag=as1de8928a
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6942dad8"
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog '8d474b1d-d46049ea-53ee6bb@172.30.0.184' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:30] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK15ba396e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.170:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK5b62726f18B0F756
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>
CSeq: 2 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4131", realm="asterisk", nonce="1a9d5d96", uri="sip:customer.com:5060", response="cd7c830536e5c751ceafe9bc8c85dcfa", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (13 headers 0 lines) ---
[2011-11-03 06:47:30] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:30] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2fcb2abc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as254e210c
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 55382de2307d742073bb4aa83ac360f4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK5b62726f18B0F756;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>;tag=as6f9f5b94
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4131@172.30.0.170:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:30 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.170' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:30] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK66dd0c7e
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1544bd08
To: "Office 107" <sip:4141@172.30.0.181:5060>;tag=5DFD67A6-E5747D
CSeq: 102 OPTIONS
Call-ID: 0b51238318aa5c995c74cb7a0f81b0a1@172.30.0.10:5060
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (14 headers 0 lines) ---
[2011-11-03 06:47:30] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK267c9de3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30] Really destroying SIP dialog '0b51238318aa5c995c74cb7a0f81b0a1@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3bac8b30
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
CSeq: 102 NOTIFY
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:30] --- (11 headers 0 lines) ---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc50f2da8A32EFFEF
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (15 headers 0 lines) ---
[2011-11-03 06:47:30] Creating new subscription
[2011-11-03 06:47:30] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:30] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:47:30] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc50f2da8A32EFFEF;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as260c65d7
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c4c85c3"
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog 'f7b8b3e-34d99d55-c99f65f4@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:30] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
Max-Forwards: 70
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
Contact: <sip:4134@172.30.0.10:5060>
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4134">
<address uri="sip:4134@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK6bd6e0bF88E7514
From: "Office 106" <sip:4120@customer.com>;tag=592E3D87-75B5FE90
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2047299d-91bcba6-94a09b2f@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (15 headers 0 lines) ---
[2011-11-03 06:47:30] Creating new subscription
[2011-11-03 06:47:30] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:30] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:30] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK6bd6e0bF88E7514;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=592E3D87-75B5FE90
To: <sip:4127@customer.com>;tag=as1047cc05
Call-ID: 2047299d-91bcba6-94a09b2f@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47263e0a"
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog '2047299d-91bcba6-94a09b2f@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:30]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2c56b55eC99118E7
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:30] --- (15 headers 0 lines) ---
[2011-11-03 06:47:30] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:30] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:30]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2c56b55eC99118E7;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>;tag=as4babbcd3
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6196ffa5"
Content-Length: 0


<------------>
[2011-11-03 06:47:30] Scheduling destruction of SIP dialog '444de9f0-e7e16879-85ffd482@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKca96fd4a52D283F1
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (15 headers 0 lines) ---
[2011-11-03 06:47:31] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:31] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKca96fd4a52D283F1;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>;tag=as536f75cd
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03cec172"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'f8c19d48-cf6d684f-190b3406@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
CSeq: 106 NOTIFY
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (11 headers 0 lines) ---
[2011-11-03 06:47:31] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:31] Retransmitting #5 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK69fb6b975DFD67A6
From: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (15 headers 0 lines) ---
[2011-11-03 06:47:31] Creating new subscription
[2011-11-03 06:47:31] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:31] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:31] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK69fb6b975DFD67A6;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as43be8f18
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1de0f4da"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.199' in 147776 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK646cb059CC3CFAAE
From: "4145" <sip:4145@customer.com>;tag=86770D49-DA6504DE
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f064cb5f-1e17ff3c-d80a02d5@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (15 headers 0 lines) ---
[2011-11-03 06:47:31] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:31] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK646cb059CC3CFAAE;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=86770D49-DA6504DE
To: <sip:4115@customer.com>;tag=as2011c671
Call-ID: f064cb5f-1e17ff3c-d80a02d5@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f67bad0"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'f064cb5f-1e17ff3c-d80a02d5@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK267c9de3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKfa669a087F11BAB
From: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (15 headers 0 lines) ---
[2011-11-03 06:47:31] Creating new subscription
[2011-11-03 06:47:31] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:31] list_route: hop: <sip:4115@172.30.0.197:5060>
[2011-11-03 06:47:31] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKfa669a087F11BAB;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2eee50f7
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6e693bd7"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'a22f4402-1ef68055-36b2a714@172.30.0.197' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKa22f44021EF68055
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (15 headers 0 lines) ---
[2011-11-03 06:47:31] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:31] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKa22f44021EF68055;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2ff9da69
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a46154f"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog '36b2a714-5b62726f-18b0f756@172.30.0.171' in 177472 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6394da8c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2fcb2abc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as254e210c
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 55382de2307d742073bb4aa83ac360f4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK8ac9d83829EBC2C1
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="6196ffa5", uri="sip:4109@customer.com:5060", response="f3fa674ed163ba09512fce59f5c064a0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (16 headers 0 lines) ---
[2011-11-03 06:47:31] Creating new subscription
[2011-11-03 06:47:31] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:31] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:31] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog '444de9f0-e7e16879-85ffd482@172.30.0.180' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK8ac9d83829EBC2C1;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
To: <sip:4109@customer.com>;tag=as4babbcd3
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:31] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:47:31] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:47:31] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1351be49
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4babbcd3
To: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.197:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK6f359e4b89F18D22
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>
CSeq: 1 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (12 headers 0 lines) ---
[2011-11-03 06:47:31] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK6f359e4b89F18D22;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>;tag=as4f450cb4
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="331f8cba"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.197' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.175:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK487347976C342DFC
From: "4140" <sip:4140@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4140@customer.com>
CSeq: 1 REGISTER
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (12 headers 0 lines) ---
[2011-11-03 06:47:31] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK487347976C342DFC;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4140@customer.com>;tag=as328acdbf
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.175
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37b3c6a1"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'e7b8eeb5-a95f4612-4fb66da3@172.30.0.175' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKba6e47326F5AB8A9
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="359ff62b", uri="sip:4148@customer.com:5060", response="fd80276399529baaf3d2545c8d1cadd0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (16 headers 0 lines) ---
[2011-11-03 06:47:31] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:31] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKba6e47326F5AB8A9;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as2406f454
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:31] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:31] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:31] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1351be49
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4babbcd3
To: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKfa669a087F11BAB
From: "Office 105" <sip:4131@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (15 headers 0 lines) ---
[2011-11-03 06:47:31] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:31] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKfa669a087F11BAB;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as6af9e4c3
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16cc0265"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'a22f4402-1ef68055-36b2a714@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK40537f6d8388F984
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="03cec172", uri="sip:4122@customer.com:5060", response="52dcded767873bacf8454cb57e5868da", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (16 headers 0 lines) ---
[2011-11-03 06:47:31] Creating new subscription
[2011-11-03 06:47:31] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:31] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:31] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog 'f8c19d48-cf6d684f-190b3406@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK40537f6d8388F984;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>;tag=as536f75cd
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:31] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:31] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:31] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK493992cc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:31]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa32effefF7B8B3E
From: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:31] --- (15 headers 0 lines) ---
[2011-11-03 06:47:31] Creating new subscription
[2011-11-03 06:47:31] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:31] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:47:31] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:31]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa32effefF7B8B3E;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as38a8c95c
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49dbbeed"
Content-Length: 0


<------------>
[2011-11-03 06:47:31] Scheduling destruction of SIP dialog '34d99d55-c99f65f4-fe46e6db@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:31] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK15ba396e
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
CSeq: 106 NOTIFY
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (11 headers 0 lines) ---
[2011-11-03 06:47:32] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1351be49
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4babbcd3
To: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa10f1dc810B9183
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (15 headers 0 lines) ---
[2011-11-03 06:47:32] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:32] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:32]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKa10f1dc810B9183;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as603feaf0
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63e41dbb"
Content-Length: 0


<------------>
[2011-11-03 06:47:32] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:32] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK493992cc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7c4af326
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7f6f144d
To: "4116" <sip:4116@172.30.0.199:5060>;tag=B769A0D1-C8120290
CSeq: 102 OPTIONS
Call-ID: 173a569a0e8c7dd35564622379d63e16@172.30.0.10:5060
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (14 headers 0 lines) ---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.175:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59c3d7e4
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1adc3c2b
To: "4140" <sip:4140@172.30.0.175:5060>;tag=EDF891B7-6F2A589C
CSeq: 102 OPTIONS
Call-ID: 6e2009487cb5947900998f6d157559ff@172.30.0.10:5060
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (14 headers 0 lines) ---
[2011-11-03 06:47:32] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK13d5a5d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as680a76d5
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f85d4c16b83b29a615cf39c6606cf56@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.172:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16db19b0
From: <sip:4134@customer.com>;tag=as675dc2fb
To: "4136" <sip:4136@customer.com>;tag=6E560E0F-F89E0928
CSeq: 106 NOTIFY
Call-ID: 17d8fc4f-2b5aa768-155b4141@172.30.0.172
Contact: <sip:4136@172.30.0.172:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (11 headers 0 lines) ---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.170:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK5b62726f18B0F756
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>
CSeq: 2 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4131", realm="asterisk", nonce="1a9d5d96", uri="sip:customer.com:5060", response="cd7c830536e5c751ceafe9bc8c85dcfa", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (13 headers 0 lines) ---
[2011-11-03 06:47:32] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:32] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0'
[2011-11-03 06:47:32]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK5b62726f18B0F756;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>;tag=as6f9f5b94
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7837c6a4", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:32] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.170' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:32] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK493992cc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2fcb2abc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as254e210c
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 55382de2307d742073bb4aa83ac360f4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
CSeq: 103 NOTIFY
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:32] --- (11 headers 0 lines) ---
[2011-11-03 06:47:32] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc50f2da8A32EFFEF
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (15 headers 0 lines) ---
[2011-11-03 06:47:32] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:32] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:32]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc50f2da8A32EFFEF;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as260c65d7
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c4c85c3"
Content-Length: 0


<------------>
[2011-11-03 06:47:32] Scheduling destruction of SIP dialog 'f7b8b3e-34d99d55-c99f65f4@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:32] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1351be49
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4babbcd3
To: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK6bd6e0bF88E7514
From: "Office 106" <sip:4120@customer.com>;tag=592E3D87-75B5FE90
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2047299d-91bcba6-94a09b2f@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (15 headers 0 lines) ---
[2011-11-03 06:47:32] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:32] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:32]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK6bd6e0bF88E7514;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=592E3D87-75B5FE90
To: <sip:4127@customer.com>;tag=as1047cc05
Call-ID: 2047299d-91bcba6-94a09b2f@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47263e0a"
Content-Length: 0


<------------>
[2011-11-03 06:47:32] Scheduling destruction of SIP dialog '2047299d-91bcba6-94a09b2f@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:32] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7658b6bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7387c248
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 707882aa2de0789f1d4896b95d62611a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:32]
<--- SIP read from UDP:172.30.0.197:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK5b62726f18B0F756
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>
CSeq: 2 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4115", realm="asterisk", nonce="331f8cba", uri="sip:customer.com:5060", response="44ccec3ed0f1f6f1cfd86a627072c0b9", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:32] --- (13 headers 0 lines) ---
[2011-11-03 06:47:32] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:32]     -- Registered SIP '4115' at 172.30.0.197:5060
[2011-11-03 06:47:32] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26115747
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6460ca59
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 516aa49e590f6eb1535d4f1b52961317@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:32]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK5b62726f18B0F756;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>;tag=as4f450cb4
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4115@172.30.0.197:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:32 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:32] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.197' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4131 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4135 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:33]   == Extension Changed 4115[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:33] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK267c9de3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2d5324ae
To: "4145" <sip:4145@customer.com>;tag=9589D1D2-2E83F7B3
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 752bdc24-7fc5859d-f2566aa2@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
CSeq: 103 NOTIFY
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:33] --- (11 headers 0 lines) ---
[2011-11-03 06:47:33] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK493992cc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 102 NOTIFY
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (11 headers 0 lines) ---
[2011-11-03 06:47:33] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKdca73ebEB92A48C
From: "Office 131" <sip:4113@customer.com>;tag=3821EBDB-B894373C
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: cf1dde01-e23d95da-bd514e7@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (15 headers 0 lines) ---
[2011-11-03 06:47:33] Creating new subscription
[2011-11-03 06:47:33] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:33] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:33] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKdca73ebEB92A48C;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=3821EBDB-B894373C
To: <sip:4108@customer.com>;tag=as35dc6a7c
Call-ID: cf1dde01-e23d95da-bd514e7@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63313934"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'cf1dde01-e23d95da-bd514e7@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:33] Retransmitting #7 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.196:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK487347976C342DFC
From: "4108" <sip:4108@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4108@customer.com>
CSeq: 1 REGISTER
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (12 headers 0 lines) ---
[2011-11-03 06:47:33] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK487347976C342DFC;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4108@customer.com>;tag=as21cbc307
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.196
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0374a33d"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'e7b8eeb5-a95f4612-4fb66da3@172.30.0.196' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:33] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK13d5a5d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as680a76d5
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f85d4c16b83b29a615cf39c6606cf56@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK40537f6d8388F984
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="03cec172", uri="sip:4122@customer.com:5060", response="52dcded767873bacf8454cb57e5868da", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (16 headers 0 lines) ---
[2011-11-03 06:47:33] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:33] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'f8c19d48-cf6d684f-190b3406@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK40537f6d8388F984;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
To: <sip:4122@customer.com>;tag=as536f75cd
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:33] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:33] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:33] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01f74fdc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20cc154d
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
CSeq: 103 NOTIFY
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:33] --- (11 headers 0 lines) ---
[2011-11-03 06:47:33] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2fcb2abc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as254e210c
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 55382de2307d742073bb4aa83ac360f4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 102 NOTIFY
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (11 headers 0 lines) ---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.187:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKc2cc960773F84E76
From: "4168" <sip:4168@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4168@customer.com>
CSeq: 1 REGISTER
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (12 headers 0 lines) ---
[2011-11-03 06:47:33] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKc2cc960773F84E76;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4168@customer.com>;tag=as66afc061
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.187
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10b37560"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.187' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:33] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01f74fdc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKa22f44021EF68055
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (15 headers 0 lines) ---
[2011-11-03 06:47:33] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:33] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKa22f44021EF68055;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2ff9da69
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a46154f"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog '36b2a714-5b62726f-18b0f756@172.30.0.171' in 177472 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa10f1dc810B9183
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (15 headers 0 lines) ---
[2011-11-03 06:47:33] Creating new subscription
[2011-11-03 06:47:33] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:33] list_route: hop: <sip:4141@172.30.0.181:5060>
[2011-11-03 06:47:33] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa10f1dc810B9183;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as7f4777df
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1609e016"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181' in 197760 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKdca73ebEB92A48C
From: "Office 131" <sip:4113@customer.com>;tag=3821EBDB-B894373C
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: cf1dde01-e23d95da-bd514e7@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (15 headers 0 lines) ---
[2011-11-03 06:47:33] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:33] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKdca73ebEB92A48C;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=3821EBDB-B894373C
To: <sip:4108@customer.com>;tag=as35dc6a7c
Call-ID: cf1dde01-e23d95da-bd514e7@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63313934"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'cf1dde01-e23d95da-bd514e7@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.175:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK487347976C342DFC
From: "4140" <sip:4140@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4140@customer.com>
CSeq: 1 REGISTER
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (12 headers 0 lines) ---
[2011-11-03 06:47:33] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK487347976C342DFC;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4140@customer.com>;tag=as328acdbf
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.175
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37b3c6a1"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'e7b8eeb5-a95f4612-4fb66da3@172.30.0.175' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:33] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6394da8c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1351be49
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4babbcd3
To: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.170:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK5b62726f18B0F756
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>
CSeq: 2 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4131", realm="asterisk", nonce="1a9d5d96", uri="sip:customer.com:5060", response="cd7c830536e5c751ceafe9bc8c85dcfa", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (13 headers 0 lines) ---
[2011-11-03 06:47:33] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:33] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0'
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK5b62726f18B0F756;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4131@customer.com>;tag=as6f9f5b94
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.170
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7837c6a4", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.170' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:33] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7658b6bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7387c248
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 707882aa2de0789f1d4896b95d62611a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:33] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01f74fdc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:33]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc50f2da8A32EFFEF
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:33] --- (15 headers 0 lines) ---
[2011-11-03 06:47:33] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:33] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:33]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc50f2da8A32EFFEF;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as260c65d7
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c4c85c3"
Content-Length: 0


<------------>
[2011-11-03 06:47:33] Scheduling destruction of SIP dialog 'f7b8b3e-34d99d55-c99f65f4@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa02f814bFE7325C2
From: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (15 headers 0 lines) ---
[2011-11-03 06:47:34] Creating new subscription
[2011-11-03 06:47:34] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:34] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:34] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:34]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa02f814bFE7325C2;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
To: <sip:4108@customer.com>;tag=as2fc941da
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61e8a2a8"
Content-Length: 0


<------------>
[2011-11-03 06:47:34] Scheduling destruction of SIP dialog 'b3f25e9-a004f0c0-87332547@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:34] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26115747
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6460ca59
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 516aa49e590f6eb1535d4f1b52961317@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:34] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37dfd4c1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63f976f6
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 180e6558110f28030bb403fd4ece1b52@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 102 NOTIFY
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (11 headers 0 lines) ---
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6394da8c
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 103 NOTIFY
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (11 headers 0 lines) ---
[2011-11-03 06:47:34] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.184:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKd5f5a5dbC92B1F80
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>
CSeq: 2 REGISTER
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4123", realm="asterisk", nonce="6942dad8", uri="sip:customer.com:5060", response="a3347718af3efd72ccbc9b543db52ba2", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (13 headers 0 lines) ---
[2011-11-03 06:47:34] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:34] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dff9779
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as334741df
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 133aa27c2cfde5eb06da0c88070cc7c2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:34]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKd5f5a5dbC92B1F80;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>;tag=as1de8928a
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4123@172.30.0.184:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:34 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:34] Scheduling destruction of SIP dialog '8d474b1d-d46049ea-53ee6bb@172.30.0.184' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:34] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK13d5a5d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as680a76d5
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f85d4c16b83b29a615cf39c6606cf56@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.178:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKc2cc960773F84E76
From: "4173" <sip:4173@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4173@customer.com>
CSeq: 1 REGISTER
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (12 headers 0 lines) ---
[2011-11-03 06:47:34] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:47:34]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKc2cc960773F84E76;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4173@customer.com>;tag=as004878f1
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.178
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59a8cf03"
Content-Length: 0


<------------>
[2011-11-03 06:47:34] Scheduling destruction of SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.178' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:34] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK493992cc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK66dd0c7e
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1544bd08
To: "Office 107" <sip:4141@172.30.0.181:5060>;tag=5DFD67A6-E5747D
CSeq: 102 OPTIONS
Call-ID: 0b51238318aa5c995c74cb7a0f81b0a1@172.30.0.10:5060
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (14 headers 0 lines) ---
[2011-11-03 06:47:34] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2fcb2abc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as254e210c
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 55382de2307d742073bb4aa83ac360f4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:34] Really destroying SIP dialog '55382de2307d742073bb4aa83ac360f4@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:34] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c42a301
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as74e7bb60
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 03a285ef58df34274ab71f2b60cd9de6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK5753c8381D26261D
From: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="31be8868", uri="sip:4122@customer.com:5060", response="e002e1c0bdad81de348e14a05dc28598", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (16 headers 0 lines) ---
[2011-11-03 06:47:34] Creating new subscription
[2011-11-03 06:47:34] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:34] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:34] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:47:34] Scheduling destruction of SIP dialog '979b6b20-c0d5d25-acdb9cce@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:34]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK5753c8381D26261D;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
To: <sip:4122@customer.com>;tag=as5ad25dc4
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:34] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:34] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:34] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6297a2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as5ad25dc4
To: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:34] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01f74fdc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.197:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK5b62726f18B0F756
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>
CSeq: 2 REGISTER
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4115", realm="asterisk", nonce="331f8cba", uri="sip:customer.com:5060", response="44ccec3ed0f1f6f1cfd86a627072c0b9", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (13 headers 0 lines) ---
[2011-11-03 06:47:34] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:34] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0'
[2011-11-03 06:47:34]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK5b62726f18B0F756;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=A1010D91-BA49E7C0
To: <sip:4115@customer.com>;tag=as4f450cb4
Call-ID: 8fcb19f5-56200334-868f8b0f@172.30.0.197
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74073daa", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:34] Scheduling destruction of SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.197' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:34] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6297a2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as5ad25dc4
To: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:34]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa10f1dc810B9183
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:34] --- (15 headers 0 lines) ---
[2011-11-03 06:47:34] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:34] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:34]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa10f1dc810B9183;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as7f4777df
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1609e016"
Content-Length: 0


<------------>
[2011-11-03 06:47:34] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181' in 197760 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:34] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7658b6bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7387c248
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 707882aa2de0789f1d4896b95d62611a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKd6a78d6a2D08A65D
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4106", realm="asterisk", nonce="5a46154f", uri="sip:4148@customer.com:5060", response="c59526c813c3ed2e34bd41ff46ab70ee", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:35] --- (16 headers 0 lines) ---
[2011-11-03 06:47:35] Creating new subscription
[2011-11-03 06:47:35] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:35] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:35] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:35] Scheduling destruction of SIP dialog '36b2a714-5b62726f-18b0f756@172.30.0.171' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:35]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKd6a78d6a2D08A65D;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2ff9da69
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:35] set_destination: Parsing <sip:4106@172.30.0.171:5060> for address/port to send to
[2011-11-03 06:47:35] set_destination: set destination to 172.30.0.171:5060
[2011-11-03 06:47:35] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3aa33491
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:35] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6297a2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as5ad25dc4
To: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:35] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26115747
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6460ca59
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 516aa49e590f6eb1535d4f1b52961317@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37dfd4c1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63f976f6
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 180e6558110f28030bb403fd4ece1b52@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKba6e47326F5AB8A9
From: "Leon Zhong" <sip:4134@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:35] --- (15 headers 0 lines) ---
[2011-11-03 06:47:35] Creating new subscription
[2011-11-03 06:47:35] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:35] list_route: hop: <sip:4134@172.30.0.179:5060>
[2011-11-03 06:47:35] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:35]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKba6e47326F5AB8A9;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as45662b56
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68dd2a9c"
Content-Length: 0


<------------>
[2011-11-03 06:47:35] Scheduling destruction of SIP dialog 'c50f2da8-a32effef-f7b8b3e@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:35] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 752bdc24-7fc5859d-f2566aa2@172.30.0.167 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6417ms with no response
[2011-11-03 06:47:35] Retransmitting #6 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:35] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dcfb75e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60a072ea
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35f4a02a2d19a4e60f1c38a61578aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35]
<--- SIP read from UDP:172.30.0.183:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK29f81f4b34544EDC
From: "4147" <sip:4147@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4147@customer.com>
CSeq: 1 REGISTER
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:35] --- (12 headers 0 lines) ---
[2011-11-03 06:47:35] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:35]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK29f81f4b34544EDC;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4147@customer.com>;tag=as780b3cce
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.183
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1bf02533"
Content-Length: 0


<------------>
[2011-11-03 06:47:35] Scheduling destruction of SIP dialog 'd595f645-e42497c6-8787fb9f@172.30.0.183' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:35] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK13d5a5d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as680a76d5
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f85d4c16b83b29a615cf39c6606cf56@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b6fe61a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1debba07
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 05e7067d343f7e9147da54de53d847a0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dff9779
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as334741df
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 133aa27c2cfde5eb06da0c88070cc7c2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6297a2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as5ad25dc4
To: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:35] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35dadc8f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as645f1b45
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 166d705e761fe5134d51fe11240d00a5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34d99d55C99F65F4
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4134", realm="asterisk", nonce="2df187d2", uri="sip:4148@customer.com:5060", response="0ce1f08d0e99a4b58f425af284160ade", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:35] --- (16 headers 0 lines) ---
[2011-11-03 06:47:35] Creating new subscription
[2011-11-03 06:47:35] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:35] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:35] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:35] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.179' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:35]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34d99d55C99F65F4;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as5e3b968c
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:35] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:47:35] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:47:35] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:35] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c42a301
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as74e7bb60
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 03a285ef58df34274ab71f2b60cd9de6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:35]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKba6e47326F5AB8A9
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:35] --- (15 headers 0 lines) ---
[2011-11-03 06:47:35] Creating new subscription
[2011-11-03 06:47:35] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:35] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:35] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:35]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKba6e47326F5AB8A9;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as48fa5dfc
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="556fec39"
Content-Length: 0


<------------>
[2011-11-03 06:47:35] Scheduling destruction of SIP dialog 'c50f2da8-a32effef-f7b8b3e@172.30.0.199' in 147776 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:35] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7658b6bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7387c248
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 707882aa2de0789f1d4896b95d62611a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:35]
<--- SIP read from UDP:172.30.0.187:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKc2cc960773F84E76
From: "4168" <sip:4168@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4168@customer.com>
CSeq: 1 REGISTER
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:35] --- (12 headers 0 lines) ---
[2011-11-03 06:47:35] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:35]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKc2cc960773F84E76;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4168@customer.com>;tag=as66afc061
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.187
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10b37560"
Content-Length: 0


<------------>
[2011-11-03 06:47:35] Scheduling destruction of SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.187' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:35] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01f74fdc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56c16298
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 102 NOTIFY
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (11 headers 0 lines) ---
[2011-11-03 06:47:36] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26115747
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6460ca59
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 516aa49e590f6eb1535d4f1b52961317@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37dfd4c1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63f976f6
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 180e6558110f28030bb403fd4ece1b52@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 752bdc24-7fc5859d-f2566aa2@172.30.0.167 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6317ms with no response
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKd6a78d6a2D08A65D
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4106", realm="asterisk", nonce="5a46154f", uri="sip:4148@customer.com:5060", response="c59526c813c3ed2e34bd41ff46ab70ee", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (16 headers 0 lines) ---
[2011-11-03 06:47:36] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:36] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog '36b2a714-5b62726f-18b0f756@172.30.0.171' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKd6a78d6a2D08A65D;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2ff9da69
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:36] set_destination: Parsing <sip:4106@172.30.0.171:5060> for address/port to send to
[2011-11-03 06:47:36] set_destination: set destination to 172.30.0.171:5060
[2011-11-03 06:47:36] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53b8ac89
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:36] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6297a2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as5ad25dc4
To: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:36] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2fcb2abc
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as254e210c
To: "Office 105" <sip:4131@172.30.0.170:5060>;tag=144ECE01-4F95F3F0
CSeq: 102 OPTIONS
Call-ID: 55382de2307d742073bb4aa83ac360f4@172.30.0.10:5060
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (14 headers 0 lines) ---
[2011-11-03 06:47:36] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dcfb75e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60a072ea
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35f4a02a2d19a4e60f1c38a61578aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK13d5a5d4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as680a76d5
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4f85d4c16b83b29a615cf39c6606cf56@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36] Really destroying SIP dialog '4f85d4c16b83b29a615cf39c6606cf56@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:36] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1351be49
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4babbcd3
To: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKb6ba9acaC4AA053
From: "Office 106" <sip:4120@customer.com>;tag=8235C922-63F652AB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2c10135c-cddf33e5-efec41ee@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (15 headers 0 lines) ---
[2011-11-03 06:47:36] Creating new subscription
[2011-11-03 06:47:36] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:36] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:36] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKb6ba9acaC4AA053;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=8235C922-63F652AB
To: <sip:4122@customer.com>;tag=as1604e1d6
Call-ID: 2c10135c-cddf33e5-efec41ee@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0cab4146"
Content-Length: 0


<------------>
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog '2c10135c-cddf33e5-efec41ee@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:36] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dff9779
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as334741df
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 133aa27c2cfde5eb06da0c88070cc7c2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKba6e47326F5AB8A9
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (15 headers 0 lines) ---
[2011-11-03 06:47:36] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:36] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKba6e47326F5AB8A9;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as48fa5dfc
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="556fec39"
Content-Length: 0


<------------>
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog 'c50f2da8-a32effef-f7b8b3e@172.30.0.199' in 147776 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa32effefF7B8B3E
From: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (15 headers 0 lines) ---
[2011-11-03 06:47:36] Creating new subscription
[2011-11-03 06:47:36] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:36] list_route: hop: <sip:4141@172.30.0.181:5060>
[2011-11-03 06:47:36] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa32effefF7B8B3E;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as6e8b7bde
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76c2b89e"
Content-Length: 0


<------------>
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog '34d99d55-c99f65f4-fe46e6db@172.30.0.181' in 197760 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:36] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b6fe61a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1debba07
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 05e7067d343f7e9147da54de53d847a0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK77964ac03AE67647
From: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (15 headers 0 lines) ---
[2011-11-03 06:47:36] Creating new subscription
[2011-11-03 06:47:36] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:36] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:47:36] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK77964ac03AE67647;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as3579c775
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3978fe44"
Content-Length: 0


<------------>
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog '347c00d6-458f082d-8eca8c0c@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:36] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35dadc8f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as645f1b45
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 166d705e761fe5134d51fe11240d00a5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.184:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKd5f5a5dbC92B1F80
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>
CSeq: 2 REGISTER
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4123", realm="asterisk", nonce="6942dad8", uri="sip:customer.com:5060", response="a3347718af3efd72ccbc9b543db52ba2", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (13 headers 0 lines) ---
[2011-11-03 06:47:36] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:36] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E'
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKd5f5a5dbC92B1F80;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=6CDEA891-8E914C6E
To: <sip:4123@customer.com>;tag=as1de8928a
Call-ID: 8d474b1d-d46049ea-53ee6bb@172.30.0.184
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="77ae55a2", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog '8d474b1d-d46049ea-53ee6bb@172.30.0.184' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:36] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c42a301
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as74e7bb60
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 03a285ef58df34274ab71f2b60cd9de6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34d99d55C99F65F4
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4134", realm="asterisk", nonce="2df187d2", uri="sip:4148@customer.com:5060", response="0ce1f08d0e99a4b58f425af284160ade", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (16 headers 0 lines) ---
[2011-11-03 06:47:36] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:36] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.179' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34d99d55C99F65F4;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as5e3b968c
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:36] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:47:36] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:47:36] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:36]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKe97f983c980B4737
From: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:36] --- (15 headers 0 lines) ---
[2011-11-03 06:47:36] Creating new subscription
[2011-11-03 06:47:36] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:36] list_route: hop: <sip:4106@172.30.0.171:5060>
[2011-11-03 06:47:36] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:36]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKe97f983c980B4737;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>;tag=as4d512a0c
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a193549"
Content-Length: 0


<------------>
[2011-11-03 06:47:36] Scheduling destruction of SIP dialog '32135a3e-144ece01-4f95f3f0@172.30.0.171' in 177472 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa32effefF7B8B3E
From: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (15 headers 0 lines) ---
[2011-11-03 06:47:37] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:37] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa32effefF7B8B3E;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as6e8b7bde
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76c2b89e"
Content-Length: 0


<------------>
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog '34d99d55-c99f65f4-fe46e6db@172.30.0.181' in 197760 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7658b6bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7387c248
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 707882aa2de0789f1d4896b95d62611a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37] Really destroying SIP dialog '707882aa2de0789f1d4896b95d62611a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:37] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26115747
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6460ca59
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 516aa49e590f6eb1535d4f1b52961317@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4115' is now UNREACHABLE!  Last qualify: 0
[2011-11-03 06:47:37] Really destroying SIP dialog '516aa49e590f6eb1535d4f1b52961317@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:47:37] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37dfd4c1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63f976f6
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 180e6558110f28030bb403fd4ece1b52@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:47:37]   == Extension Changed 4115[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:47:37] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK493992cc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKfe46e6db1C0E3CA
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="3c4c85c3", uri="sip:4109@customer.com:5060", response="3eddbcff52c1df27b4fccd649a96969a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (16 headers 0 lines) ---
[2011-11-03 06:47:37] Creating new subscription
[2011-11-03 06:47:37] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:37] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:37] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog 'f7b8b3e-34d99d55-c99f65f4@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKfe46e6db1C0E3CA;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as260c65d7
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:37] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:37] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:37] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKd6a78d6a2D08A65D
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4106", realm="asterisk", nonce="5a46154f", uri="sip:4148@customer.com:5060", response="c59526c813c3ed2e34bd41ff46ab70ee", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (16 headers 0 lines) ---
[2011-11-03 06:47:37] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:37] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog '36b2a714-5b62726f-18b0f756@172.30.0.171' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKd6a78d6a2D08A65D;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2ff9da69
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:37] set_destination: Parsing <sip:4106@172.30.0.171:5060> for address/port to send to
[2011-11-03 06:47:37] set_destination: set destination to 172.30.0.171:5060
[2011-11-03 06:47:37] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08076f7c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37] Retransmitting #8 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dcfb75e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60a072ea
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35f4a02a2d19a4e60f1c38a61578aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4c48df21
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as383f2e3a
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 510e72012c6b73b81d5bc2e44c4e40c8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dff9779
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as334741df
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 133aa27c2cfde5eb06da0c88070cc7c2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6394da8c
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 103 NOTIFY
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (11 headers 0 lines) ---
[2011-11-03 06:47:37] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b6fe61a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1debba07
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 05e7067d343f7e9147da54de53d847a0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35dadc8f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as645f1b45
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 166d705e761fe5134d51fe11240d00a5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK8ad6943220CAE52B
From: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (15 headers 0 lines) ---
[2011-11-03 06:47:37] Creating new subscription
[2011-11-03 06:47:37] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:37] list_route: hop: <sip:4147@172.30.0.183:5060>
[2011-11-03 06:47:37] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK8ad6943220CAE52B;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
To: <sip:4148@customer.com>;tag=as32698f53
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c88c375"
Content-Length: 0


<------------>
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog '198759bc-5af6fe25-2cc894a6@172.30.0.183' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKb50ece3b778761DA
From: "4168" <sip:4168@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (15 headers 0 lines) ---
[2011-11-03 06:47:37] Creating new subscription
[2011-11-03 06:47:37] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:37] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:47:37] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKb50ece3b778761DA;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>;tag=as7298f0e0
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.187
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a6ea22f"
Content-Length: 0


<------------>
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog '78031bfd-6cc68e94-cf2353cf@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c42a301
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as74e7bb60
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 03a285ef58df34274ab71f2b60cd9de6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKba6e47326F5AB8A9
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (15 headers 0 lines) ---
[2011-11-03 06:47:37] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:37] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKba6e47326F5AB8A9;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as48fa5dfc
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="556fec39"
Content-Length: 0


<------------>
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog 'c50f2da8-a32effef-f7b8b3e@172.30.0.199' in 147776 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa10f1dc810B9183
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (15 headers 0 lines) ---
[2011-11-03 06:47:37] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:37] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKa10f1dc810B9183;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as7f4777df
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1609e016"
Content-Length: 0


<------------>
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181' in 197760 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37] Really destroying SIP dialog 'f064cb5f-1e17ff3c-d80a02d5@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:47:37] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6297a2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as5ad25dc4
To: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41857c2b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36fc1329
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 22b3bfeb5c84c7716a93710635a43b71@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:37] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3aa33491
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:37]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4149@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK2884e7e989157F16
From: "4108" <sip:4108@customer.com>;tag=5112CAB4-B941C04D
To: <sip:4149@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: edf891b7-6f2a589c-89545bd5@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:37] --- (15 headers 0 lines) ---
[2011-11-03 06:47:37] Creating new subscription
[2011-11-03 06:47:37] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:37] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:47:37] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:47:37]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK2884e7e989157F16;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=5112CAB4-B941C04D
To: <sip:4149@customer.com>;tag=as3bb3c09f
Call-ID: edf891b7-6f2a589c-89545bd5@172.30.0.196
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="77e55889"
Content-Length: 0


<------------>
[2011-11-03 06:47:37] Scheduling destruction of SIP dialog 'edf891b7-6f2a589c-89545bd5@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:37] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47d34405
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a3b9d6c
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0025ecf14f1cef4869bb67166ec37a01@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK1bd4e18cC43602E7
From: "4173" <sip:4173@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (15 headers 0 lines) ---
[2011-11-03 06:47:38] Creating new subscription
[2011-11-03 06:47:38] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:47:38] list_route: hop: <sip:4173@172.30.0.178:5060>
[2011-11-03 06:47:38] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:47:38]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK1bd4e18cC43602E7;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as6666be01
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.178
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="62df66f9"
Content-Length: 0


<------------>
[2011-11-03 06:47:38] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKd712de335E4C9262
From: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4124", realm="asterisk", nonce="49dbbeed", uri="sip:4109@customer.com:5060", response="d7342f9cf93ebacb44c40c662dc48b9f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (16 headers 0 lines) ---
[2011-11-03 06:47:38] Creating new subscription
[2011-11-03 06:47:38] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:38] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:38] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:38] Scheduling destruction of SIP dialog '34d99d55-c99f65f4-fe46e6db@172.30.0.164' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:38]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKd712de335E4C9262;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as38a8c95c
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:38] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:47:38] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:47:38] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:38] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37dfd4c1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as63f976f6
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 180e6558110f28030bb403fd4ece1b52@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38] Really destroying SIP dialog '180e6558110f28030bb403fd4ece1b52@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:38] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1351be49
From: <sip:4109@customer.com>;tag=as4babbcd3
To: "Office 106" <sip:4120@customer.com>;tag=5A48B6C-DEE9C3F5
CSeq: 102 NOTIFY
Call-ID: 444de9f0-e7e16879-85ffd482@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (11 headers 0 lines) ---
[2011-11-03 06:47:38] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:38] Really destroying SIP dialog 'a22f4402-1ef68055-36b2a714@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:47:38] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dcfb75e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60a072ea
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35f4a02a2d19a4e60f1c38a61578aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK66cb7d776D712680
From: "Office 106" <sip:4120@customer.com>;tag=42BDEE3D-C7CD8046
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 35317d09-75c0c112-fd33329b@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (15 headers 0 lines) ---
[2011-11-03 06:47:38] Creating new subscription
[2011-11-03 06:47:38] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:38] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:38] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:38]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK66cb7d776D712680;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=42BDEE3D-C7CD8046
To: <sip:4108@customer.com>;tag=as1ea3947e
Call-ID: 35317d09-75c0c112-fd33329b@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e895934"
Content-Length: 0


<------------>
[2011-11-03 06:47:38] Scheduling destruction of SIP dialog '35317d09-75c0c112-fd33329b@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:38] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dff9779
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as334741df
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 133aa27c2cfde5eb06da0c88070cc7c2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38] Really destroying SIP dialog '133aa27c2cfde5eb06da0c88070cc7c2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:38] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4c48df21
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as383f2e3a
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 510e72012c6b73b81d5bc2e44c4e40c8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b6fe61a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1debba07
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 05e7067d343f7e9147da54de53d847a0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35dadc8f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as645f1b45
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 166d705e761fe5134d51fe11240d00a5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c42a301
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as74e7bb60
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 03a285ef58df34274ab71f2b60cd9de6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:38] Really destroying SIP dialog '03a285ef58df34274ab71f2b60cd9de6@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.166:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKb1e2adab9542CE66
From: "4137" <sip:4137@customer.com>;tag=971A06C5-F38B6420
To: <sip:4137@customer.com>
CSeq: 1 REGISTER
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (12 headers 0 lines) ---
[2011-11-03 06:47:38] Sending to 172.30.0.166:5060 (no NAT)
[2011-11-03 06:47:38]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKb1e2adab9542CE66;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=971A06C5-F38B6420
To: <sip:4137@customer.com>;tag=as6517afda
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.166
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="331db230"
Content-Length: 0


<------------>
[2011-11-03 06:47:38] Scheduling destruction of SIP dialog '3b0be151-a656316c-99d39b7@172.30.0.166' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:38] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01f74fdc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.176:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK29f81f4b34544EDC
From: "4122" <sip:4122@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4122@customer.com>
CSeq: 1 REGISTER
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.176
Contact: <sip:4122@172.30.0.176:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (12 headers 0 lines) ---
[2011-11-03 06:47:38] Sending to 172.30.0.176:5060 (no NAT)
[2011-11-03 06:47:38]
<--- Transmitting (no NAT) to 172.30.0.176:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK29f81f4b34544EDC;received=172.30.0.176
From: "4122" <sip:4122@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4122@customer.com>;tag=as34d205a0
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.176
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ad0cc82"
Content-Length: 0


<------------>
[2011-11-03 06:47:38] Scheduling destruction of SIP dialog 'd595f645-e42497c6-8787fb9f@172.30.0.176' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK9d139879F0B4E548
From: "Office 105" <sip:4131@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 4c6094f3-d6a78d6a-2d08a65d@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (15 headers 0 lines) ---
[2011-11-03 06:47:38] Creating new subscription
[2011-11-03 06:47:38] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:38] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:47:38] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:47:38]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK9d139879F0B4E548;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>;tag=as41955856
Call-ID: 4c6094f3-d6a78d6a-2d08a65d@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a25c239"
Content-Length: 0


<------------>
[2011-11-03 06:47:38] Scheduling destruction of SIP dialog '4c6094f3-d6a78d6a-2d08a65d@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:38]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK77964ac03AE67647
From: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:38] --- (15 headers 0 lines) ---
[2011-11-03 06:47:38] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:38] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:38]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK77964ac03AE67647;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as3579c775
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3978fe44"
Content-Length: 0


<------------>
[2011-11-03 06:47:38] Scheduling destruction of SIP dialog '347c00d6-458f082d-8eca8c0c@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:38] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41857c2b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36fc1329
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 22b3bfeb5c84c7716a93710635a43b71@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:39]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKb50ece3b778761DA
From: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:39] --- (15 headers 0 lines) ---
[2011-11-03 06:47:39] Creating new subscription
[2011-11-03 06:47:39] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:39] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:47:39] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:39]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKb50ece3b778761DA;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>;tag=as20a415b7
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4da15ab2"
Content-Length: 0


<------------>
[2011-11-03 06:47:39] Scheduling destruction of SIP dialog '78031bfd-6cc68e94-cf2353cf@172.30.0.168' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:39] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47d34405
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a3b9d6c
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0025ecf14f1cef4869bb67166ec37a01@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:39] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53b8ac89
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK295b7265A601323
From: "Office 131" <sip:4113@customer.com>;tag=1E83FF0A-5425C1D7
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: df1ce224-c5b03179-e464acb2@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:39] --- (15 headers 0 lines) ---
[2011-11-03 06:47:39] Creating new subscription
[2011-11-03 06:47:39] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:39] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:39] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:39]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK295b7265A601323;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=1E83FF0A-5425C1D7
To: <sip:4115@customer.com>;tag=as6450ae65
Call-ID: df1ce224-c5b03179-e464acb2@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3bf4346d"
Content-Length: 0


<------------>
[2011-11-03 06:47:39] Scheduling destruction of SIP dialog 'df1ce224-c5b03179-e464acb2@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:39] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54f244bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10eebce8
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a8509fc3c5ce7fc5667856d27edac3b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:39] Retransmitting #7 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK34d99d55C99F65F4
From: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="1de0f4da", uri="sip:4148@customer.com:5060", response="db1a26d660a0917b8ca09f329c090557", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:39] --- (16 headers 0 lines) ---
[2011-11-03 06:47:39] Creating new subscription
[2011-11-03 06:47:39] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:39] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:39] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:39] Scheduling destruction of SIP dialog 'e5747d-a10f1dc-810b9183@172.30.0.199' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:39]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK34d99d55C99F65F4;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as43be8f18
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:39] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:47:39] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:47:39] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1834367a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dcfb75e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as60a072ea
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35f4a02a2d19a4e60f1c38a61578aee3@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:39] Really destroying SIP dialog '35f4a02a2d19a4e60f1c38a61578aee3@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:39] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKe97f983c980B4737
From: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="6e693bd7", uri="sip:4148@customer.com:5060", response="c641653f6b08f74d493bf819cedc8faf", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:39] --- (16 headers 0 lines) ---
[2011-11-03 06:47:39] Creating new subscription
[2011-11-03 06:47:39] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:39] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:47:39] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:39] Scheduling destruction of SIP dialog 'a22f4402-1ef68055-36b2a714@172.30.0.197' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:39]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKe97f983c980B4737;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
To: <sip:4148@customer.com>;tag=as2eee50f7
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:39] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:47:39] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:47:39] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56987e84
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2eee50f7
To: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4c48df21
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as383f2e3a
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 510e72012c6b73b81d5bc2e44c4e40c8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:39]
<--- SIP read from UDP:172.30.0.165:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK93490dc318E53540
From: "Office 113" <sip:4171@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4171@customer.com>
CSeq: 1 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:39] --- (12 headers 0 lines) ---
[2011-11-03 06:47:39] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:47:39]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK93490dc318E53540;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4171@customer.com>;tag=as77058084
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.165
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="037998e5"
Content-Length: 0


<------------>
[2011-11-03 06:47:39] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.165' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:39] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b6fe61a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1debba07
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 05e7067d343f7e9147da54de53d847a0@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:39] Really destroying SIP dialog '05e7067d343f7e9147da54de53d847a0@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:39] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56987e84
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2eee50f7
To: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35dadc8f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as645f1b45
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 166d705e761fe5134d51fe11240d00a5@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:39] Really destroying SIP dialog '166d705e761fe5134d51fe11240d00a5@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:39]
<--- SIP read from UDP:172.30.0.168:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKc2cc960773F84E76
From: "4125" <sip:4125@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4125@customer.com>
CSeq: 1 REGISTER
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:39] --- (12 headers 0 lines) ---
[2011-11-03 06:47:39] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:39]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKc2cc960773F84E76;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4125@customer.com>;tag=as6a271104
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.168
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5592c282"
Content-Length: 0


<------------>
[2011-11-03 06:47:39] Scheduling destruction of SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.168' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:39] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56987e84
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2eee50f7
To: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:39]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK59c9257fB7400EF0
From: "4147" <sip:4147@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:39] --- (15 headers 0 lines) ---
[2011-11-03 06:47:40] Creating new subscription
[2011-11-03 06:47:40] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:40] list_route: hop: <sip:4147@172.30.0.183:5060>
[2011-11-03 06:47:40] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK59c9257fB7400EF0;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>;tag=as4da6a684
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.183
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="663ebd6d"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog '7fd09539-8cc4f29a-6f06e753@172.30.0.183' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:40] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41857c2b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36fc1329
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 22b3bfeb5c84c7716a93710635a43b71@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47d34405
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a3b9d6c
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0025ecf14f1cef4869bb67166ec37a01@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.163:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK93490dc318E53540
From: "4138" <sip:4138@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4138@customer.com>
CSeq: 1 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (12 headers 0 lines) ---
[2011-11-03 06:47:40] Sending to 172.30.0.163:5060 (no NAT)
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK93490dc318E53540;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4138@customer.com>;tag=as7cb1c5ac
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.163
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68d26b54"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.163' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:40] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6cf64638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as14c2f1e0
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7ab260576b5e9e0e61298d847c7b0f02@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK197fee5947D4D9D8
From: "4124" <sip:4124@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 89034e9f-25f4486e-db6f3505@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (15 headers 0 lines) ---
[2011-11-03 06:47:40] Creating new subscription
[2011-11-03 06:47:40] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:40] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:47:40] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK197fee5947D4D9D8;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as35b1eb30
Call-ID: 89034e9f-25f4486e-db6f3505@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="093ab3ad"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog '89034e9f-25f4486e-db6f3505@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:40] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56987e84
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2eee50f7
To: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:40] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08076f7c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKd712de335E4C9262
From: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4124", realm="asterisk", nonce="49dbbeed", uri="sip:4109@customer.com:5060", response="d7342f9cf93ebacb44c40c662dc48b9f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (16 headers 0 lines) ---
[2011-11-03 06:47:40] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:40] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog '34d99d55-c99f65f4-fe46e6db@172.30.0.164' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKd712de335E4C9262;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as38a8c95c
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:40] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:47:40] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:47:40] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a9912bb
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:40] Really destroying SIP dialog 'cf1dde01-e23d95da-bd514e7@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK694818368891BFBF
From: "Office 106" <sip:4120@customer.com>;tag=1296CD8-F3BE4761
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f4c3d4c8-d2329751-4b92475a@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (15 headers 0 lines) ---
[2011-11-03 06:47:40] Creating new subscription
[2011-11-03 06:47:40] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:40] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:40] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK694818368891BFBF;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=1296CD8-F3BE4761
To: <sip:4115@customer.com>;tag=as3534b2c6
Call-ID: f4c3d4c8-d2329751-4b92475a@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54839435"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog 'f4c3d4c8-d2329751-4b92475a@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:40] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54f244bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10eebce8
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a8509fc3c5ce7fc5667856d27edac3b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40] Retransmitting #5 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.187:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKaba210a5F75BE89C
From: "4168" <sip:4168@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4168@customer.com>
CSeq: 2 REGISTER
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4168", realm="asterisk", nonce="10b37560", uri="sip:customer.com:5060", response="8ca9b08704f40c85b733732bdff58c84", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (13 headers 0 lines) ---
[2011-11-03 06:47:40] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:40] NOTICE[9661]: chan_sip.c:25638 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2011-11-03 06:47:40] Really destroying SIP dialog '510e72012c6b73b81d5bc2e44c4e40c8@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:40] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52e9e6f0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b2d3700
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 680d489229e2da3f32050d303a040a80@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKaba210a5F75BE89C;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4168@customer.com>;tag=as66afc061
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.187
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4168@172.30.0.187:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:40 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.187' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:40] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:40] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK1bd4e18cC43602E7
From: "4168" <sip:4168@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (15 headers 0 lines) ---
[2011-11-03 06:47:40] Creating new subscription
[2011-11-03 06:47:40] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:40] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:47:40] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK1bd4e18cC43602E7;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as001cb61e
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.187
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a586252"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:40] Reliably Transmitting (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42dc6715
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as339ae45c
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 79076f0d5d17a2845c067c0413fdc5d7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK59c9257fB7400EF0
From: "4147" <sip:4147@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (15 headers 0 lines) ---
[2011-11-03 06:47:40] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:40] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK59c9257fB7400EF0;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>;tag=as4da6a684
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.183
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="663ebd6d"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog '7fd09539-8cc4f29a-6f06e753@172.30.0.183' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:40] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e21fb80
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69ad8930
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f8ca1d3465f5cde38992e7e51d5fafd@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKdd948a9fF0C43E50
From: "Office 131" <sip:4113@customer.com>;tag=3821EBDB-B894373C
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: cf1dde01-e23d95da-bd514e7@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="63313934", uri="sip:4108@customer.com:5060", response="0e983d415d50c4b952df406d4635ac85", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (16 headers 0 lines) ---
[2011-11-03 06:47:40] Creating new subscription
[2011-11-03 06:47:40] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:40] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:40] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKdd948a9fF0C43E50;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=3821EBDB-B894373C
To: <sip:4108@customer.com>;tag=as78833302
Call-ID: cf1dde01-e23d95da-bd514e7@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56abb2cd"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog 'cf1dde01-e23d95da-bd514e7@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:40] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41857c2b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36fc1329
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 22b3bfeb5c84c7716a93710635a43b71@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47d34405
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a3b9d6c
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0025ecf14f1cef4869bb67166ec37a01@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:40] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56987e84
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2eee50f7
To: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:40]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK197fee5947D4D9D8
From: "4124" <sip:4124@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 89034e9f-25f4486e-db6f3505@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:40] --- (15 headers 0 lines) ---
[2011-11-03 06:47:40] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:40] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:40]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK197fee5947D4D9D8;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as35b1eb30
Call-ID: 89034e9f-25f4486e-db6f3505@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="093ab3ad"
Content-Length: 0


<------------>
[2011-11-03 06:47:40] Scheduling destruction of SIP dialog '89034e9f-25f4486e-db6f3505@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:41] Retransmitting #6 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6297a2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as5ad25dc4
To: "Office 131" <sip:4113@customer.com>;tag=9D5694D0-526BF495
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 979b6b20-c0d5d25-acdb9cce@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41]
<--- SIP read from UDP:172.30.0.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7658b6bf
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7387c248
To: "4138" <sip:4138@172.30.0.163:5060>;tag=3F880E63-4C393B60
CSeq: 102 OPTIONS
Call-ID: 707882aa2de0789f1d4896b95d62611a@172.30.0.10:5060
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:41] --- (14 headers 0 lines) ---
[2011-11-03 06:47:41] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 979b6b20-c0d5d25-acdb9cce@172.30.0.195 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6414ms with no response
[2011-11-03 06:47:41] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a9912bb
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKe9b9a2a9B618F20E
From: "Office 113" <sip:4171@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 9708841f-46781d2c-65ddf445@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:41] --- (15 headers 0 lines) ---
[2011-11-03 06:47:41] Creating new subscription
[2011-11-03 06:47:41] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:47:41] list_route: hop: <sip:4171@172.30.0.165:5060>
[2011-11-03 06:47:41] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:47:41]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKe9b9a2a9B618F20E;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>;tag=as030a296b
Call-ID: 9708841f-46781d2c-65ddf445@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69553763"
Content-Length: 0


<------------>
[2011-11-03 06:47:41] Scheduling destruction of SIP dialog '9708841f-46781d2c-65ddf445@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:41] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK493992cc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6cf64638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as14c2f1e0
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7ab260576b5e9e0e61298d847c7b0f02@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:41]
<--- SIP read from UDP:172.30.0.177:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK93490dc318E53540
From: "4135" <sip:4135@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4135@customer.com>
CSeq: 1 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:41] --- (12 headers 0 lines) ---
[2011-11-03 06:47:41] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:47:41]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK93490dc318E53540;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4135@customer.com>;tag=as19312c1e
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.177
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51918e05"
Content-Length: 0


<------------>
[2011-11-03 06:47:41] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.177' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:41] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54f244bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10eebce8
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a8509fc3c5ce7fc5667856d27edac3b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:41] Retransmitting #9 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK84fda8269853BA89
From: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:41] --- (15 headers 0 lines) ---
[2011-11-03 06:47:41] Creating new subscription
[2011-11-03 06:47:41] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:41] list_route: hop: <sip:4106@172.30.0.171:5060>
[2011-11-03 06:47:41] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:41]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK84fda8269853BA89;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>;tag=as539e8ced
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30b8aaa3"
Content-Length: 0


<------------>
[2011-11-03 06:47:41] Scheduling destruction of SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.171' in 177472 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:41] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK458f082d8ECA8C0C
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="1609e016", uri="sip:4148@customer.com:5060", response="597ce11aca05c3a4e95d45f7d3fe00dd", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:41] --- (16 headers 0 lines) ---
[2011-11-03 06:47:41] Creating new subscription
[2011-11-03 06:47:41] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:41] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:41] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:41] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:41]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK458f082d8ECA8C0C;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as7f4777df
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:41] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:47:41] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:47:41] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19f6d50b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKfe46e6db1C0E3CA
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="3c4c85c3", uri="sip:4109@customer.com:5060", response="3eddbcff52c1df27b4fccd649a96969a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:41] --- (16 headers 0 lines) ---
[2011-11-03 06:47:41] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:41] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:41] Scheduling destruction of SIP dialog 'f7b8b3e-34d99d55-c99f65f4@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:41]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKfe46e6db1C0E3CA;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as260c65d7
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:41] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:41] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:41] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK605cfdb1
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52e9e6f0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b2d3700
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 680d489229e2da3f32050d303a040a80@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:41] Retransmitting #1 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42dc6715
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as339ae45c
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 79076f0d5d17a2845c067c0413fdc5d7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:41] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1834367a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e21fb80
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69ad8930
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f8ca1d3465f5cde38992e7e51d5fafd@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:41] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKfe46e6db1C0E3CA
From: "Leon Zhong" <sip:4134@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 43c45281-77964ac0-3ae67647@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:41] --- (15 headers 0 lines) ---
[2011-11-03 06:47:41] Creating new subscription
[2011-11-03 06:47:41] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:41] list_route: hop: <sip:4134@172.30.0.179:5060>
[2011-11-03 06:47:41] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:41]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKfe46e6db1C0E3CA;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as45548058
Call-ID: 43c45281-77964ac0-3ae67647@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59e40645"
Content-Length: 0


<------------>
[2011-11-03 06:47:41] Scheduling destruction of SIP dialog '43c45281-77964ac0-3ae67647@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:41] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3aa33491
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:41] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41857c2b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36fc1329
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 22b3bfeb5c84c7716a93710635a43b71@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:41] Really destroying SIP dialog '22b3bfeb5c84c7716a93710635a43b71@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:41] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47d34405
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a3b9d6c
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0025ecf14f1cef4869bb67166ec37a01@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:41] Really destroying SIP dialog '0025ecf14f1cef4869bb67166ec37a01@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:42]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1bd4e18cC43602E7
From: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:42] --- (15 headers 0 lines) ---
[2011-11-03 06:47:42] Creating new subscription
[2011-11-03 06:47:42] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:42] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:47:42] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:42]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1bd4e18cC43602E7;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as7b42bfd5
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ed8ef53"
Content-Length: 0


<------------>
[2011-11-03 06:47:42] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.168' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:42] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK605cfdb1
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:42] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6cf64638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as14c2f1e0
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7ab260576b5e9e0e61298d847c7b0f02@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:42]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKd536ad1e6322A631
From: "4168" <sip:4168@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:42] --- (15 headers 0 lines) ---
[2011-11-03 06:47:42] Creating new subscription
[2011-11-03 06:47:42] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:42] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:47:42] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:47:42]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKd536ad1e6322A631;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>;tag=as37cea88c
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.187
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="325e1851"
Content-Length: 0


<------------>
[2011-11-03 06:47:42] Scheduling destruction of SIP dialog '8303abf8-652cc7a3-461beda2@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:42] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54f244bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10eebce8
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a8509fc3c5ce7fc5667856d27edac3b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:42]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK5069c9a4364A15CD
From: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:42] --- (15 headers 0 lines) ---
[2011-11-03 06:47:42] Creating new subscription
[2011-11-03 06:47:42] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:42] list_route: hop: <sip:4147@172.30.0.183:5060>
[2011-11-03 06:47:42] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:47:42]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK5069c9a4364A15CD;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>;tag=as17dabd16
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16b80892"
Content-Length: 0


<------------>
[2011-11-03 06:47:42] Scheduling destruction of SIP dialog 'cfe67e0e-c5367aa7-166259d8@172.30.0.183' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:42]
<--- SIP read from UDP:172.30.0.163:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK93490dc318E53540
From: "4138" <sip:4138@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4138@customer.com>
CSeq: 1 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:42] --- (12 headers 0 lines) ---
[2011-11-03 06:47:42] Sending to 172.30.0.163:5060 (no NAT)
[2011-11-03 06:47:42]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK93490dc318E53540;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4138@customer.com>;tag=as7cb1c5ac
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.163
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68d26b54"
Content-Length: 0


<------------>
[2011-11-03 06:47:42] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.163' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:42]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK21f724e33CD56FEC
From: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:42] --- (15 headers 0 lines) ---
[2011-11-03 06:47:42] Creating new subscription
[2011-11-03 06:47:42] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:42] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:42] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:42]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK21f724e33CD56FEC;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
To: <sip:4147@customer.com>;tag=as07c04e0f
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="266c3f90"
Content-Length: 0


<------------>
[2011-11-03 06:47:42] Scheduling destruction of SIP dialog '3a016875-fdaf4e7e-42736207@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:42] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission f8c19d48-cf6d684f-190b3406@172.30.0.173 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10573ms with no response
[2011-11-03 06:47:42]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK458f082d8ECA8C0C
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="1609e016", uri="sip:4148@customer.com:5060", response="597ce11aca05c3a4e95d45f7d3fe00dd", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:42] --- (16 headers 0 lines) ---
[2011-11-03 06:47:42] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:42] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:42] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:42]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK458f082d8ECA8C0C;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as7f4777df
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:42] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:47:42] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:47:42] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:42] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52e9e6f0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b2d3700
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 680d489229e2da3f32050d303a040a80@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:42] Retransmitting #5 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56987e84
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2eee50f7
To: "4115" <sip:4115@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a22f4402-1ef68055-36b2a714@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:42] Retransmitting #2 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42dc6715
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as339ae45c
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 79076f0d5d17a2845c067c0413fdc5d7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:42] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK605cfdb1
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:42] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK01f74fdc
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as536f75cd
To: "4118" <sip:4118@customer.com>;tag=A851D332-F446E59
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f8c19d48-cf6d684f-190b3406@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:42] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a9912bb
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:42]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK829899a23F880E63
From: "4135" <sip:4135@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 4c393b60-e9b9a2a9-b618f20e@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:42] --- (15 headers 0 lines) ---
[2011-11-03 06:47:42] Creating new subscription
[2011-11-03 06:47:42] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:47:42] list_route: hop: <sip:4135@172.30.0.177:5060>
[2011-11-03 06:47:42] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:47:42]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK829899a23F880E63;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>;tag=as26cc82d5
Call-ID: 4c393b60-e9b9a2a9-b618f20e@172.30.0.177
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="779f6eac"
Content-Length: 0


<------------>
[2011-11-03 06:47:42] Scheduling destruction of SIP dialog '4c393b60-e9b9a2a9-b618f20e@172.30.0.177' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:42] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e21fb80
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69ad8930
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f8ca1d3465f5cde38992e7e51d5fafd@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:42] Really destroying SIP dialog '2047299d-91bcba6-94a09b2f@172.30.0.180' Method: SUBSCRIBE
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK347c00d6458F082D
From: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (15 headers 0 lines) ---
[2011-11-03 06:47:43] Creating new subscription
[2011-11-03 06:47:43] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:43] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:43] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:43]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK347c00d6458F082D;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as2eff3dab
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41dff001"
Content-Length: 0


<------------>
[2011-11-03 06:47:43] Scheduling destruction of SIP dialog '8eca8c0c-d712de33-5e4c9262@172.30.0.199' in 147776 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:43] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53b8ac89
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK197fee5947D4D9D8
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="556fec39", uri="sip:4109@customer.com:5060", response="fb2a3d0d2016d7f5259db75e63710980", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (16 headers 0 lines) ---
[2011-11-03 06:47:43] Creating new subscription
[2011-11-03 06:47:43] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:43] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:43] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:43] Scheduling destruction of SIP dialog 'c50f2da8-a32effef-f7b8b3e@172.30.0.199' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:43]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK197fee5947D4D9D8;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as48fa5dfc
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:43] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:47:43] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:47:43] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK1c0e3ca43C45281
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4124", realm="asterisk", nonce="63e41dbb", uri="sip:4148@customer.com:5060", response="8cac13402b81004f0df58d9cc881e77e", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (16 headers 0 lines) ---
[2011-11-03 06:47:43] Creating new subscription
[2011-11-03 06:47:43] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:43] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:43] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:43] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:43]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK1c0e3ca43C45281;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as603feaf0
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:43] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:47:43] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:47:43] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6cf64638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as14c2f1e0
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7ab260576b5e9e0e61298d847c7b0f02@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:43] Retransmitting #8 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK26115747
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6460ca59
To: "4115" <sip:4115@172.30.0.197:5060>;tag=144ECE01-4F95F3F0
CSeq: 102 OPTIONS
Call-ID: 516aa49e590f6eb1535d4f1b52961317@172.30.0.10:5060
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (14 headers 0 lines) ---
[2011-11-03 06:47:43] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54f244bf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10eebce8
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3a8509fc3c5ce7fc5667856d27edac3b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:43] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4116' is now UNREACHABLE!  Last qualify: 2309
[2011-11-03 06:47:43] Really destroying SIP dialog '3a8509fc3c5ce7fc5667856d27edac3b@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKafdfcf0d7CBC6752
From: "4145" <sip:4145@customer.com>;tag=B6C579AB-89AD5B98
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 4ed85f33-99e3da80-72a000c9@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (15 headers 0 lines) ---
[2011-11-03 06:47:43] Creating new subscription
[2011-11-03 06:47:43] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:43] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:43] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:43]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKafdfcf0d7CBC6752;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=B6C579AB-89AD5B98
To: <sip:4119@customer.com>;tag=as31c6719c
Call-ID: 4ed85f33-99e3da80-72a000c9@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="684fd3f8"
Content-Length: 0


<------------>
[2011-11-03 06:47:43] Scheduling destruction of SIP dialog '4ed85f33-99e3da80-72a000c9@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:43] Really destroying SIP dialog '4814d2fa-ee5ed3ad-4e3bf144@172.30.0.189' Method: REGISTER
[2011-11-03 06:47:43] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52e9e6f0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b2d3700
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 680d489229e2da3f32050d303a040a80@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK13d5a5d4
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as680a76d5
To: "Office 113" <sip:4171@172.30.0.165:5060>;tag=3F880E63-4C393B60
CSeq: 102 OPTIONS
Call-ID: 4f85d4c16b83b29a615cf39c6606cf56@172.30.0.10:5060
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (14 headers 0 lines) ---
[2011-11-03 06:47:43] Retransmitting #3 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42dc6715
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as339ae45c
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 79076f0d5d17a2845c067c0413fdc5d7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3dff9779
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as334741df
To: "4123" <sip:4123@172.30.0.184:5060>;tag=A99174D4-E2CD37ED
CSeq: 102 OPTIONS
Call-ID: 133aa27c2cfde5eb06da0c88070cc7c2@172.30.0.10:5060
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (14 headers 0 lines) ---
[2011-11-03 06:47:43] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e21fb80
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69ad8930
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f8ca1d3465f5cde38992e7e51d5fafd@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK197fee5947D4D9D8
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="556fec39", uri="sip:4109@customer.com:5060", response="fb2a3d0d2016d7f5259db75e63710980", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (16 headers 0 lines) ---
[2011-11-03 06:47:43] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:43] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:43] Scheduling destruction of SIP dialog 'c50f2da8-a32effef-f7b8b3e@172.30.0.199' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:43]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK197fee5947D4D9D8;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as48fa5dfc
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:43] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:47:43] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:47:43] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d88f9ec
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.196:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2b6fe61a
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1debba07
To: "4108" <sip:4108@172.30.0.196:5060>;tag=4740DAA-6B0F1D9B
CSeq: 102 OPTIONS
Call-ID: 05e7067d343f7e9147da54de53d847a0@172.30.0.10:5060
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (14 headers 0 lines) ---
[2011-11-03 06:47:43] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK605cfdb1
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK8eca8c0cD712DE33
From: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (15 headers 0 lines) ---
[2011-11-03 06:47:43] Creating new subscription
[2011-11-03 06:47:43] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:43] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:47:43] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:43]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK8eca8c0cD712DE33;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as44540955
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="73a6a1e8"
Content-Length: 0


<------------>
[2011-11-03 06:47:43] Scheduling destruction of SIP dialog '5e4c9262-197fee59-47d4d9d8@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:43]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c42a301
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as74e7bb60
To: "4173" <sip:4173@172.30.0.178:5060>;tag=652CC7A3-461BEDA2
CSeq: 102 OPTIONS
Call-ID: 03a285ef58df34274ab71f2b60cd9de6@172.30.0.10:5060
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:43] --- (14 headers 0 lines) ---
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.176:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK6060cfe1E09E0702
From: "4122" <sip:4122@customer.com>;tag=8ADDFE22-C406D99B
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2ff02f7b-7c7d8f8c-62981375@172.30.0.176
Contact: <sip:4122@172.30.0.176:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (15 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.176:5060 (no NAT)
[2011-11-03 06:47:44] list_route: hop: <sip:4122@172.30.0.176:5060>
[2011-11-03 06:47:44] Found peer '4122' for '4122' from 172.30.0.176:5060
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.176:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK6060cfe1E09E0702;received=172.30.0.176
From: "4122" <sip:4122@customer.com>;tag=8ADDFE22-C406D99B
To: <sip:4122@customer.com>;tag=as6652e086
Call-ID: 2ff02f7b-7c7d8f8c-62981375@172.30.0.176
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60dcb9a4"
Content-Length: 0


<------------>
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog '2ff02f7b-7c7d8f8c-62981375@172.30.0.176' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Really destroying SIP dialog '78031bfd-6cc68e94-cf2353cf@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:47:44] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission f8c19d48-cf6d684f-190b3406@172.30.0.173 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10651ms with no response
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK43ae4a40F60B7ACB
From: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4106", realm="asterisk", nonce="7a193549", uri="sip:4109@customer.com:5060", response="2f6ebc6bf72e17510be1fc9a4f8402fb", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (16 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:44] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:44] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog '32135a3e-144ece01-4f95f3f0@172.30.0.171' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK43ae4a40F60B7ACB;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>;tag=as4d512a0c
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:44] set_destination: Parsing <sip:4106@172.30.0.171:5060> for address/port to send to
[2011-11-03 06:47:44] set_destination: set destination to 172.30.0.171:5060
[2011-11-03 06:47:44] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6cf64638
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as14c2f1e0
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7ab260576b5e9e0e61298d847c7b0f02@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:44] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4106' is now UNREACHABLE!  Last qualify: 2773
[2011-11-03 06:47:44] Really destroying SIP dialog '7ab260576b5e9e0e61298d847c7b0f02@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa5760e3f3ADADCF6
From: "4118" <sip:4118@customer.com>;tag=1BD55451-10F09BA8
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 85339b5d-941df874-9f68d33b@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (15 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:44] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:44] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa5760e3f3ADADCF6;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=1BD55451-10F09BA8
To: <sip:4147@customer.com>;tag=as59e06266
Call-ID: 85339b5d-941df874-9f68d33b@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4249893a"
Content-Length: 0


<------------>
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog '85339b5d-941df874-9f68d33b@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08076f7c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Retransmitting #6 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK567d512b4B5FDD18
From: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (15 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:44] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:44] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK567d512b4B5FDD18;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
To: <sip:4138@customer.com>;tag=as72d45b6b
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d321885"
Content-Length: 0


<------------>
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog 'd4966781-a4f38ab6-81a022c7@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44] Really destroying SIP dialog 'edf891b7-6f2a589c-89545bd5@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK458f082d8ECA8C0C
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="1609e016", uri="sip:4148@customer.com:5060", response="597ce11aca05c3a4e95d45f7d3fe00dd", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (16 headers 0 lines) ---
[2011-11-03 06:47:44] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:44] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog 'ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK458f082d8ECA8C0C;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
To: <sip:4148@customer.com>;tag=as7f4777df
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:44] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:47:44] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:47:44] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7fe241bb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Really destroying SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.178' Method: SUBSCRIBE
[2011-11-03 06:47:44] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK1153d83a5A70009B
From: "Office 113" <sip:4171@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (15 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:47:44] list_route: hop: <sip:4171@172.30.0.165:5060>
[2011-11-03 06:47:44] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK1153d83a5A70009B;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as19371b3a
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60894ea7"
Content-Length: 0


<------------>
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog 'ddf9dab8-ab9de2a1-75ab5826@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52e9e6f0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b2d3700
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 680d489229e2da3f32050d303a040a80@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:44] Really destroying SIP dialog '680d489229e2da3f32050d303a040a80@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:44] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19f6d50b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK44e6b00e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26362b71
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71a7bec35cb1fe352d8b8ab41fdea5d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:44] Retransmitting #4 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42dc6715
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as339ae45c
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 79076f0d5d17a2845c067c0413fdc5d7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:44] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4169' is now UNREACHABLE!  Last qualify: 69
[2011-11-03 06:47:44] Really destroying SIP dialog '79076f0d5d17a2845c067c0413fdc5d7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:44] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:44] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:44] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7994e92b
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as2d7b6337
To: "Office 131" <sip:4113@customer.com>;tag=5FABFDD6-8510CD8D
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: bfee6331-b25fc778-66739a8f@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44]   == Extension Changed 4169[default-local] new state Unavailable for Notify User 4113
[2011-11-03 06:47:44] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:44] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:44] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f3554c2
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44]   == Extension Changed 4169[default-local] new state Unavailable for Notify User 4113
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKd536ad1e6322A631
From: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (15 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:44] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:47:44] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKd536ad1e6322A631;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>;tag=as7283c46d
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3768806d"
Content-Length: 0


<------------>
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog '8303abf8-652cc7a3-461beda2@172.30.0.168' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44] Really destroying SIP dialog 'a8664db1-74778406-262df01b@172.30.0.173' Method: SUBSCRIBE
[2011-11-03 06:47:44] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e21fb80
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as69ad8930
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f8ca1d3465f5cde38992e7e51d5fafd@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:44] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4141' is now UNREACHABLE!  Last qualify: 3090
[2011-11-03 06:47:44] Really destroying SIP dialog '1f8ca1d3465f5cde38992e7e51d5fafd@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK5c5c927e370F3765
From: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="61e8a2a8", uri="sip:4108@customer.com:5060", response="2118a0de44143d370fed278463a94377", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (16 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:44] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:44] Looking for 4108 in default-local (domain customer.com:5060)
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog 'b3f25e9-a004f0c0-87332547@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK5c5c927e370F3765;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
To: <sip:4108@customer.com>;tag=as2fc941da
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4108@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:44] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:44] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:44] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3b3e3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2fc941da
To: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7994e92b
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as2d7b6337
To: "Office 131" <sip:4113@customer.com>;tag=5FABFDD6-8510CD8D
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: bfee6331-b25fc778-66739a8f@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f3554c2
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK24ce725e34E0994F
From: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:44] --- (15 headers 0 lines) ---
[2011-11-03 06:47:44] Creating new subscription
[2011-11-03 06:47:44] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:44] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:44] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:44]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK24ce725e34E0994F;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
To: <sip:4140@customer.com>;tag=as46091936
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f5e92fb"
Content-Length: 0


<------------>
[2011-11-03 06:47:44] Scheduling destruction of SIP dialog '2645a66c-fc9dca45-6725fa2a@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:44] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7994e92b
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as2d7b6337
To: "Office 131" <sip:4113@customer.com>;tag=5FABFDD6-8510CD8D
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: bfee6331-b25fc778-66739a8f@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:44] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f3554c2
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKbd552310E296DB19
From: "4108" <sip:4108@customer.com>;tag=1B63D766-80BD747
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 973fd306-61221967-8836730c@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (15 headers 0 lines) ---
[2011-11-03 06:47:45] Creating new subscription
[2011-11-03 06:47:45] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:45] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:47:45] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKbd552310E296DB19;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=1B63D766-80BD747
To: <sip:4148@customer.com>;tag=as529b4e57
Call-ID: 973fd306-61221967-8836730c@172.30.0.196
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50f1db1f"
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog '973fd306-61221967-8836730c@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:45] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3b3e3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2fc941da
To: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK1bd4e18cC43602E7
From: "4173" <sip:4173@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (15 headers 0 lines) ---
[2011-11-03 06:47:45] Creating new subscription
[2011-11-03 06:47:45] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:47:45] list_route: hop: <sip:4173@172.30.0.178:5060>
[2011-11-03 06:47:45] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK1bd4e18cC43602E7;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as0c51767d
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.178
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6392951d"
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKaba210a5F75BE89C
From: "4173" <sip:4173@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="62df66f9", uri="sip:4148@customer.com:5060", response="3a4bea867af25b1ad657dd93e5f6a1eb", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (16 headers 0 lines) ---
[2011-11-03 06:47:45] Creating new subscription
[2011-11-03 06:47:45] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:47:45] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:47:45] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4173" <sip:4173@customer.com>;tag=AF6D521D-30E74C34'
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKaba210a5F75BE89C;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as0c51767d
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.178
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6392951d", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:45] Really destroying SIP dialog '4c6094f3-d6a78d6a-2d08a65d@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKb4cfe424A9E0F78B
From: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4124", realm="asterisk", nonce="3978fe44", uri="sip:4127@customer.com:5060", response="807cac34dfa63889b4543b37636f6774", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (16 headers 0 lines) ---
[2011-11-03 06:47:45] Creating new subscription
[2011-11-03 06:47:45] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:45] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:45] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog '347c00d6-458f082d-8eca8c0c@172.30.0.164' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKb4cfe424A9E0F78B;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as3579c775
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:45] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:47:45] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:47:45] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Retransmitting #10 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.182:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK29f81f4b34544EDC
From: "Office 108" <sip:4128@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4128@customer.com>
CSeq: 1 REGISTER
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (12 headers 0 lines) ---
[2011-11-03 06:47:45] Sending to 172.30.0.182:5060 (no NAT)
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK29f81f4b34544EDC;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4128@customer.com>;tag=as7fe7f883
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.182
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="556799c2"
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog 'd595f645-e42497c6-8787fb9f@172.30.0.182' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:45] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7994e92b
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as2d7b6337
To: "Office 131" <sip:4113@customer.com>;tag=5FABFDD6-8510CD8D
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: bfee6331-b25fc778-66739a8f@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f3554c2
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3b3e3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2fc941da
To: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1bd4e18cC43602E7
From: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (15 headers 0 lines) ---
[2011-11-03 06:47:45] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:45] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1bd4e18cC43602E7;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as7b42bfd5
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ed8ef53"
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.168' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK89034e9f25F4486E
From: "4116" <sip:4116@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (15 headers 0 lines) ---
[2011-11-03 06:47:45] Creating new subscription
[2011-11-03 06:47:45] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:45] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:45] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bK89034e9f25F4486E;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>;tag=as63930fb4
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ef1682c"
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog 'db6f3505-b4cfe424-a9e0f78b@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:45] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK44e6b00e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26362b71
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71a7bec35cb1fe352d8b8ab41fdea5d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:45] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1834367a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Really destroying SIP dialog 'df1ce224-c5b03179-e464acb2@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.188:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKfeb328ecC390509B
From: "4149" <sip:4149@customer.com>;tag=C9B5A5D6-69D51C95
To: <sip:4149@customer.com>
CSeq: 1 REGISTER
Call-ID: d9e96f62-e46aa701-9213cf38@172.30.0.188
Contact: <sip:4149@172.30.0.188:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (12 headers 0 lines) ---
[2011-11-03 06:47:45] Sending to 172.30.0.188:5060 (no NAT)
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.188:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKfeb328ecC390509B;received=172.30.0.188
From: "4149" <sip:4149@customer.com>;tag=C9B5A5D6-69D51C95
To: <sip:4149@customer.com>;tag=as10e31326
Call-ID: d9e96f62-e46aa701-9213cf38@172.30.0.188
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5424f303"
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog 'd9e96f62-e46aa701-9213cf38@172.30.0.188' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:45] Retransmitting #5 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a22f4402-1ef68055-36b2a714@172.30.0.197 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6386ms with no response
[2011-11-03 06:47:45] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK295b7265A601323
From: "Office 131" <sip:4113@customer.com>;tag=1E83FF0A-5425C1D7
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: df1ce224-c5b03179-e464acb2@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (15 headers 0 lines) ---
[2011-11-03 06:47:45] Creating new subscription
[2011-11-03 06:47:45] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:45] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:45] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:45]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK295b7265A601323;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=1E83FF0A-5425C1D7
To: <sip:4115@customer.com>;tag=as447f7816
Call-ID: df1ce224-c5b03179-e464acb2@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3551a774"
Content-Length: 0


<------------>
[2011-11-03 06:47:45] Scheduling destruction of SIP dialog 'df1ce224-c5b03179-e464acb2@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:45] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3aa33491
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a9912bb
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:45]
<--- SIP read from UDP:172.30.0.176:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK41857c2b
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36fc1329
To: "4122" <sip:4122@172.30.0.176:5060>;tag=BEF255CF-13453AC0
CSeq: 102 OPTIONS
Call-ID: 22b3bfeb5c84c7716a93710635a43b71@172.30.0.10:5060
Contact: <sip:4122@172.30.0.176:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:45] --- (14 headers 0 lines) ---
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.183:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK6060cfe1E09E0702
From: "4147" <sip:4147@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4147@customer.com>
CSeq: 2 REGISTER
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4147", realm="asterisk", nonce="1bf02533", uri="sip:customer.com:5060", response="ef03c8b0289aa25142841b8496d16a87", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (13 headers 0 lines) ---
[2011-11-03 06:47:46] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:46]     -- Registered SIP '4147' at 172.30.0.183:5060
[2011-11-03 06:47:46] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76ab7578
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as096c908b
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35e9902d35cd8ea864f4dc4a709f3f6a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK6060cfe1E09E0702;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4147@customer.com>;tag=as780b3cce
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.183
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4147@172.30.0.183:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:46 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4131 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4136 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4137 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4147[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog 'd595f645-e42497c6-8787fb9f@172.30.0.183' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:46] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d88f9ec
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:46] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3b3e3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2fc941da
To: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKbf5b42fc7B0A1443
From: "4118" <sip:4118@customer.com>;tag=584FA26C-3A6FAEB3
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 52853a3a-7c6d73e1-2ee1b038@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (15 headers 0 lines) ---
[2011-11-03 06:47:46] Creating new subscription
[2011-11-03 06:47:46] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:46] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:46] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKbf5b42fc7B0A1443;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=584FA26C-3A6FAEB3
To: <sip:4115@customer.com>;tag=as750ffb8f
Call-ID: 52853a3a-7c6d73e1-2ee1b038@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="094bb6b6"
Content-Length: 0


<------------>
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog '52853a3a-7c6d73e1-2ee1b038@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:46] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:46] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7994e92b
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as2d7b6337
To: "Office 131" <sip:4113@customer.com>;tag=5FABFDD6-8510CD8D
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: bfee6331-b25fc778-66739a8f@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:46] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f3554c2
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.176:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bKdd39308922C0516A
From: "4122" <sip:4122@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4122@customer.com>
CSeq: 2 REGISTER
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.176
Contact: <sip:4122@172.30.0.176:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4122", realm="asterisk", nonce="1ad0cc82", uri="sip:customer.com:5060", response="9ec3f310a5f45f814c8429d416c63a4f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (13 headers 0 lines) ---
[2011-11-03 06:47:46] Sending to 172.30.0.176:5060 (no NAT)
[2011-11-03 06:47:46] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK626093b8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b7fc95f
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1e025d894d55f530573b086334348435@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.176:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bKdd39308922C0516A;received=172.30.0.176
From: "4122" <sip:4122@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4122@customer.com>;tag=as34d205a0
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.176
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.176:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:46 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog 'd595f645-e42497c6-8787fb9f@172.30.0.176' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:46] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK605cfdb1
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:46] Really destroying SIP dialog '2c10135c-cddf33e5-efec41ee@172.30.0.180' Method: SUBSCRIBE
[2011-11-03 06:47:46] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6c704847
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as75fa06d6
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4fec6db71c9ef3c8566e27785f30cc73@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK8c749db795D6C766
From: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (15 headers 0 lines) ---
[2011-11-03 06:47:46] Creating new subscription
[2011-11-03 06:47:46] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:47:46] list_route: hop: <sip:4173@172.30.0.178:5060>
[2011-11-03 06:47:46] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK8c749db795D6C766;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>;tag=as1ce2bd26
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5df1d7a9"
Content-Length: 0


<------------>
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog 'a02e6f59-4d240880-e0e88a0b@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK8ad6943220CAE52B
From: "Office 108" <sip:4128@customer.com>;tag=88DB6959-26E871BA
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (15 headers 0 lines) ---
[2011-11-03 06:47:46] Creating new subscription
[2011-11-03 06:47:46] Sending to 172.30.0.182:5060 (no NAT)
[2011-11-03 06:47:46] list_route: hop: <sip:4128@172.30.0.182:5060>
[2011-11-03 06:47:46] Found peer '4128' for '4128' from 172.30.0.182:5060
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK8ad6943220CAE52B;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=88DB6959-26E871BA
To: <sip:4148@customer.com>;tag=as74010d52
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.182
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="138b08f2"
Content-Length: 0


<------------>
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog '198759bc-5af6fe25-2cc894a6@172.30.0.182' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.168:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1e6a23acCCD99C87
From: "4125" <sip:4125@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4125@customer.com>
CSeq: 2 REGISTER
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Authorization: Digest username="4125", realm="asterisk", nonce="5592c282", uri="sip:customer.com:5060", response="d9c64d499417b198b8aa3979cc330738", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (13 headers 0 lines) ---
[2011-11-03 06:47:46] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:46]     -- Registered SIP '4125' at 172.30.0.168:5060
[2011-11-03 06:47:46] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3806872b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76b1d284
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60a65ea60b162e467a2d787f63ec3ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1e6a23acCCD99C87;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4125@customer.com>;tag=as6a271104
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.168
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4125@172.30.0.168:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:46 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4131 (queued)
[2011-11-03 06:47:46] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:47:46] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:47:46] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4136
[2011-11-03 06:47:46]   == Extension Changed 4125[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.168' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:46] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK44e6b00e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26362b71
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71a7bec35cb1fe352d8b8ab41fdea5d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK6d5f5fd7BF46E004
From: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (15 headers 0 lines) ---
[2011-11-03 06:47:46] Creating new subscription
[2011-11-03 06:47:46] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:47:46] list_route: hop: <sip:4171@172.30.0.165:5060>
[2011-11-03 06:47:46] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK6d5f5fd7BF46E004;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>;tag=as159032dc
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="64d27540"
Content-Length: 0


<------------>
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog 'd2df29bd-7b13dd2-3f183dd3@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:46]
<--- SIP read from UDP:172.30.0.169:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.169:5060;branch=z9hG4bK93490dc318E53540
From: "4127" <sip:4127@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4127@customer.com>
CSeq: 1 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.169
Contact: <sip:4127@172.30.0.169:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:46] --- (12 headers 0 lines) ---
[2011-11-03 06:47:46] Sending to 172.30.0.169:5060 (no NAT)
[2011-11-03 06:47:46]
<--- Transmitting (no NAT) to 172.30.0.169:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.169:5060;branch=z9hG4bK93490dc318E53540;received=172.30.0.169
From: "4127" <sip:4127@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4127@customer.com>;tag=as3a50f2aa
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.169
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45ab85f9"
Content-Length: 0


<------------>
[2011-11-03 06:47:46] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.169' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:46] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKc1452592C5F32C09
From: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (15 headers 0 lines) ---
[2011-11-03 06:47:47] Creating new subscription
[2011-11-03 06:47:47] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:47] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:47] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKc1452592C5F32C09;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as21c64c30
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4806050c"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog '76ce4e08-4f93254f-34c24d9e@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47] Really destroying SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:47:47] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53b8ac89
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.165:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK507559109D8E4599
From: "Office 113" <sip:4171@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4171@customer.com>
CSeq: 2 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4171", realm="asterisk", nonce="037998e5", uri="sip:customer.com:5060", response="7730bcb72f6e725bce8deecbafd29137", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (13 headers 0 lines) ---
[2011-11-03 06:47:47] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:47:47] NOTICE[9661]: chan_sip.c:25638 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2011-11-03 06:47:47] Really destroying SIP dialog '4fec6db71c9ef3c8566e27785f30cc73@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:47] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a32cc84
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b1aef55
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 472d4757150639851e7cbcf93968e4b8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK507559109D8E4599;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4171@customer.com>;tag=as77058084
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.165
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4171@172.30.0.165:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:47 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.165' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:47] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76ab7578
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as096c908b
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35e9902d35cd8ea864f4dc4a709f3f6a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:47] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a2db5cf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1102b6dd
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 663e81e1654cded60e01fdf44e33daa1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:47] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b930f17
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6c549c
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3993488f7a3e144950e29ff625fd2e42@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.168:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1e6a23acCCD99C87
From: "4125" <sip:4125@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4125@customer.com>
CSeq: 2 REGISTER
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Authorization: Digest username="4125", realm="asterisk", nonce="5592c282", uri="sip:customer.com:5060", response="d9c64d499417b198b8aa3979cc330738", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (13 headers 0 lines) ---
[2011-11-03 06:47:47] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:47] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4125" <sip:4125@customer.com>;tag=713B5A75-A9C9FB2C'
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK1e6a23acCCD99C87;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=713B5A75-A9C9FB2C
To: <sip:4125@customer.com>;tag=as6a271104
Call-ID: f41d8a29-20eba810-5722db5b@172.30.0.168
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2256400e", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.168' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:47] Retransmitting #9 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK1153d83a5A70009B
From: "Office 113" <sip:4171@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (15 headers 0 lines) ---
[2011-11-03 06:47:47] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:47] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK1153d83a5A70009B;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as19371b3a
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60894ea7"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog 'ddf9dab8-ab9de2a1-75ab5826@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK4d240880E0E88A0B
From: "4125" <sip:4125@customer.com>;tag=E07B4379-52A6EC20
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 22da1e6a-a2bb164d-aa9e0fa4@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (15 headers 0 lines) ---
[2011-11-03 06:47:47] Creating new subscription
[2011-11-03 06:47:47] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:47] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:47:47] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK4d240880E0E88A0B;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=E07B4379-52A6EC20
To: <sip:4108@customer.com>;tag=as39e9ec46
Call-ID: 22da1e6a-a2bb164d-aa9e0fa4@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05e7bd4e"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog '22da1e6a-a2bb164d-aa9e0fa4@172.30.0.168' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47] Really destroying SIP dialog 'cf1dde01-e23d95da-bd514e7@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:47:47] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3b3e3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2fc941da
To: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKd536ad1e6322A631
From: "4168" <sip:4168@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (15 headers 0 lines) ---
[2011-11-03 06:47:47] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:47] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKd536ad1e6322A631;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>;tag=as37cea88c
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.187
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="325e1851"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog '8303abf8-652cc7a3-461beda2@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK626093b8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b7fc95f
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1e025d894d55f530573b086334348435@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKc716d2fa72870C31
From: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (15 headers 0 lines) ---
[2011-11-03 06:47:47] Creating new subscription
[2011-11-03 06:47:47] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:47] list_route: hop: <sip:4134@172.30.0.179:5060>
[2011-11-03 06:47:47] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKc716d2fa72870C31;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as771d3887
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ba5e4a7"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK43c4528177964AC0
From: "empty station" <sip:4119@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3ae67647-347c00d6-458f082d@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (15 headers 0 lines) ---
[2011-11-03 06:47:47] Creating new subscription
[2011-11-03 06:47:47] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:47] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:47:47] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK43c4528177964AC0;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as619115b2
Call-ID: 3ae67647-347c00d6-458f082d@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="173f3cfa"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog '3ae67647-347c00d6-458f082d@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7fe241bb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a9912bb
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
CSeq: 103 NOTIFY
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (11 headers 0 lines) ---
[2011-11-03 06:47:47] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:47] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK44e6b00e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26362b71
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71a7bec35cb1fe352d8b8ab41fdea5d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:47] Really destroying SIP dialog '9708841f-46781d2c-65ddf445@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.176:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK59c9257fB7400EF0
From: "4122" <sip:4122@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.176
Contact: <sip:4122@172.30.0.176:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (15 headers 0 lines) ---
[2011-11-03 06:47:47] Creating new subscription
[2011-11-03 06:47:47] Sending to 172.30.0.176:5060 (no NAT)
[2011-11-03 06:47:47] list_route: hop: <sip:4122@172.30.0.176:5060>
[2011-11-03 06:47:47] Found peer '4122' for '4122' from 172.30.0.176:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.176:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK59c9257fB7400EF0;received=172.30.0.176
From: "4122" <sip:4122@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>;tag=as39466b87
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.176
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="361b6788"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog '7fd09539-8cc4f29a-6f06e753@172.30.0.176' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK1646577a4A3B43B
From: "4145" <sip:4145@customer.com>;tag=86770D49-DA6504DE
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f064cb5f-1e17ff3c-d80a02d5@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="6f67bad0", uri="sip:4115@customer.com:5060", response="fcd9781dc39f6ed127351085bd7e6310", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (16 headers 0 lines) ---
[2011-11-03 06:47:47] Creating new subscription
[2011-11-03 06:47:47] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:47] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:47] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK1646577a4A3B43B;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=86770D49-DA6504DE
To: <sip:4115@customer.com>;tag=as7d10ae59
Call-ID: f064cb5f-1e17ff3c-d80a02d5@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="782cbcae"
Content-Length: 0


<------------>
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog 'f064cb5f-1e17ff3c-d80a02d5@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47] Retransmitting #7 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3806872b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76b1d284
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60a65ea60b162e467a2d787f63ec3ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:47] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7994e92b
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as2d7b6337
To: "Office 131" <sip:4113@customer.com>;tag=5FABFDD6-8510CD8D
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: bfee6331-b25fc778-66739a8f@172.30.0.195
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4f3554c2
Max-Forwards: 70
From: <sip:4169@customer.com>;tag=as34506259
To: "Office 131" <sip:4113@customer.com>;tag=CA6B930C-86070E81
Contact: <sip:4169@172.30.0.10:5060>
Call-ID: 330f082c-b8c37221-ae9a447a@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4169">
<address uri="sip:4169@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:47]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK9e85010a5D40047D
From: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4106", realm="asterisk", nonce="30b8aaa3", uri="sip:4122@customer.com:5060", response="2870e36521daf627cd4842bd0b8b55ae", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:47] --- (16 headers 0 lines) ---
[2011-11-03 06:47:47] Creating new subscription
[2011-11-03 06:47:47] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:47] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:47] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:47:47] Scheduling destruction of SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.171' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:47]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK9e85010a5D40047D;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>;tag=as539e8ced
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:47] set_destination: Parsing <sip:4106@172.30.0.171:5060> for address/port to send to
[2011-11-03 06:47:47] set_destination: set destination to 172.30.0.171:5060
[2011-11-03 06:47:47] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.184:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKf1a7a342B0C80AF3
From: "4123" <sip:4123@customer.com>;tag=61CA9FD3-D8EB30D8
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 16a0ba78-aedee661-2e9fbe7e@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (15 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:48] list_route: hop: <sip:4123@172.30.0.184:5060>
[2011-11-03 06:47:48] Found peer '4123' for '4123' from 172.30.0.184:5060
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKf1a7a342B0C80AF3;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=61CA9FD3-D8EB30D8
To: <sip:4127@customer.com>;tag=as698f6868
Call-ID: 16a0ba78-aedee661-2e9fbe7e@172.30.0.184
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72a2ca1d"
Content-Length: 0


<------------>
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog '16a0ba78-aedee661-2e9fbe7e@172.30.0.184' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.166:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK251567bc3586D07
From: "4137" <sip:4137@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e56f2582-f4496c2d-8cab4c08@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (15 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.166:5060 (no NAT)
[2011-11-03 06:47:48] list_route: hop: <sip:4137@172.30.0.166:5060>
[2011-11-03 06:47:48] Found peer '4137' for '4137' from 172.30.0.166:5060
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK251567bc3586D07;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>;tag=as20cae31d
Call-ID: e56f2582-f4496c2d-8cab4c08@172.30.0.166
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3061e589"
Content-Length: 0


<------------>
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog 'e56f2582-f4496c2d-8cab4c08@172.30.0.166' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76ab7578
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as096c908b
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35e9902d35cd8ea864f4dc4a709f3f6a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.184:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bK453b209d3DCDCD6A
From: "4123" <sip:4123@customer.com>;tag=4CBED857-6DFD0BAC
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e42bb03b-4ddb3960-32efff69@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (15 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:48] list_route: hop: <sip:4123@172.30.0.184:5060>
[2011-11-03 06:47:48] Found peer '4123' for '4123' from 172.30.0.184:5060
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bK453b209d3DCDCD6A;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=4CBED857-6DFD0BAC
To: <sip:4147@customer.com>;tag=as72c28b10
Call-ID: e42bb03b-4ddb3960-32efff69@172.30.0.184
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a7f955a"
Content-Length: 0


<------------>
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog 'e42bb03b-4ddb3960-32efff69@172.30.0.184' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00741eda
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2acd3794
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4bb193d547bf5a492e52b25b08ab30a4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a32cc84
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b1aef55
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 472d4757150639851e7cbcf93968e4b8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a2db5cf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1102b6dd
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 663e81e1654cded60e01fdf44e33daa1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b930f17
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6c549c
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3993488f7a3e144950e29ff625fd2e42@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08076f7c
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK35a1e4c6DCF0644F
From: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (15 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:48] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:48] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK35a1e4c6DCF0644F;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
To: <sip:4119@customer.com>;tag=as773db5d8
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1aa894ea"
Content-Length: 0


<------------>
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog '136b5158-e366ebe1-5d9773ea@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48] Retransmitting #7 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4eae1528
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKc716d2fa72870C31
From: "4116" <sip:4116@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (15 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:48] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:48] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKc716d2fa72870C31;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as498b1507
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dba1b29"
Content-Length: 0


<------------>
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48] Really destroying SIP dialog '35317d09-75c0c112-fd33329b@172.30.0.180' Method: SUBSCRIBE
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK673c37316CEA51D2
From: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4147", realm="asterisk", nonce="16b80892", uri="sip:4127@customer.com:5060", response="7149e01ae2a95df04fd505625b06d99c", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (16 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:48] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:47:48] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog 'cfe67e0e-c5367aa7-166259d8@172.30.0.183' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK673c37316CEA51D2;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>;tag=as17dabd16
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:48] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:47:48] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:47:48] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK626093b8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b7fc95f
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1e025d894d55f530573b086334348435@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.184:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKb7a3a009589E2946
From: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (15 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:48] list_route: hop: <sip:4123@172.30.0.184:5060>
[2011-11-03 06:47:48] Found peer '4123' for '4123' from 172.30.0.184:5060
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKb7a3a009589E2946;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
To: <sip:4109@customer.com>;tag=as1bf9566b
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53d536ce"
Content-Length: 0


<------------>
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog '6b074147-5962c75c-3c206d15@172.30.0.184' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19f6d50b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16274433
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36e753a2
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3e18f430bd400f654bff9128151289@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK44e6b00e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as26362b71
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71a7bec35cb1fe352d8b8ab41fdea5d9@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Really destroying SIP dialog '71a7bec35cb1fe352d8b8ab41fdea5d9@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:48] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ec2b685
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as17f6477a
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f36c00a40aa5e3274b5842251035611@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2911297313484C7C
From: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="266c3f90", uri="sip:4147@customer.com:5060", response="a89338d66127d82e6993de76772ce4f5", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (16 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:48] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:48] Looking for 4147 in default-local (domain customer.com:5060)
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog '3a016875-fdaf4e7e-42736207@172.30.0.180' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2911297313484C7C;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
To: <sip:4147@customer.com>;tag=as07c04e0f
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4147@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:48] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:47:48] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:47:48] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48329534
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as07c04e0f
To: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3806872b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76b1d284
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60a65ea60b162e467a2d787f63ec3ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:48] Really destroying SIP dialog 'bf918fb-34b59fea-1750b009@172.30.0.190' Method: REGISTER
[2011-11-03 06:47:48]
<--- SIP read from UDP:172.30.0.192:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK831dfc1541EBC670
From: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:48] --- (15 headers 0 lines) ---
[2011-11-03 06:47:48] Creating new subscription
[2011-11-03 06:47:48] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:47:48] list_route: hop: <sip:4148@172.30.0.192:5060>
[2011-11-03 06:47:48] Found peer '4148' for '4148' from 172.30.0.192:5060
[2011-11-03 06:47:48]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK831dfc1541EBC670;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>;tag=as252ea80a
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1bdc9853"
Content-Length: 0


<------------>
[2011-11-03 06:47:48] Scheduling destruction of SIP dialog 'cccbccfb-9f3dfab6-424f6aa1@172.30.0.192' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:48] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:48] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48329534
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as07c04e0f
To: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK46589a73EBF68D48
From: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (15 headers 0 lines) ---
[2011-11-03 06:47:49] Creating new subscription
[2011-11-03 06:47:49] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:49] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:47:49] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:47:49]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK46589a73EBF68D48;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4109@customer.com>;tag=as53f2e80f
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08488643"
Content-Length: 0


<------------>
[2011-11-03 06:47:49] Scheduling destruction of SIP dialog '834d5031-6928867e-29412cbf@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:49] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76ab7578
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as096c908b
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35e9902d35cd8ea864f4dc4a709f3f6a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53b8ac89
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
CSeq: 103 NOTIFY
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:49] --- (11 headers 0 lines) ---
[2011-11-03 06:47:49] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:49] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a32cc84
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b1aef55
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 472d4757150639851e7cbcf93968e4b8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKaba210a5F75BE89C
From: "4125" <sip:4125@customer.com>;tag=67D2CA3C-121DD8D7
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 8c749db7-95d6c766-a02e6f59@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (15 headers 0 lines) ---
[2011-11-03 06:47:49] Creating new subscription
[2011-11-03 06:47:49] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:49] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:47:49] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:49]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKaba210a5F75BE89C;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=67D2CA3C-121DD8D7
To: <sip:4122@customer.com>;tag=as1f3b2157
Call-ID: 8c749db7-95d6c766-a02e6f59@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78da1f2a"
Content-Length: 0


<------------>
[2011-11-03 06:47:49] Scheduling destruction of SIP dialog '8c749db7-95d6c766-a02e6f59@172.30.0.168' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:49] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a2db5cf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1102b6dd
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 663e81e1654cded60e01fdf44e33daa1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b930f17
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6c549c
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3993488f7a3e144950e29ff625fd2e42@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48329534
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as07c04e0f
To: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00741eda
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2acd3794
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4bb193d547bf5a492e52b25b08ab30a4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #11 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK43ae4a40F60B7ACB
From: "4115" <sip:4115@customer.com>;tag=E84B1746-E4C25229
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: b7988ba2-27b08275-7d13db4@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (15 headers 0 lines) ---
[2011-11-03 06:47:49] Creating new subscription
[2011-11-03 06:47:49] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:49] list_route: hop: <sip:4115@172.30.0.197:5060>
[2011-11-03 06:47:49] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:47:49]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK43ae4a40F60B7ACB;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=E84B1746-E4C25229
To: <sip:4115@customer.com>;tag=as52fa7299
Call-ID: b7988ba2-27b08275-7d13db4@172.30.0.197
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52e44a83"
Content-Length: 0


<------------>
[2011-11-03 06:47:49] Scheduling destruction of SIP dialog 'b7988ba2-27b08275-7d13db4@172.30.0.197' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:49] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ab3e09e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as04208b25
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78216a6342355e9c5111f1265af3f94c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK626093b8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b7fc95f
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1e025d894d55f530573b086334348435@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54f244bf
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as10eebce8
To: "4116" <sip:4116@172.30.0.199:5060>;tag=6F0CEE3C-8442B2E3
CSeq: 102 OPTIONS
Call-ID: 3a8509fc3c5ce7fc5667856d27edac3b@172.30.0.10:5060
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (14 headers 0 lines) ---
[2011-11-03 06:47:49] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK89034e9f25F4486E
From: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (15 headers 0 lines) ---
[2011-11-03 06:47:49] Creating new subscription
[2011-11-03 06:47:49] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:49] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:47:49] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:49]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK89034e9f25F4486E;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>;tag=as3a063028
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42c0d266"
Content-Length: 0


<------------>
[2011-11-03 06:47:49] Scheduling destruction of SIP dialog 'db6f3505-b4cfe424-a9e0f78b@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.186:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42dc6715
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as339ae45c
To: "4169" <sip:4169@172.30.0.186:5060>;tag=2DAEF199-E2917A2C
CSeq: 102 OPTIONS
Call-ID: 79076f0d5d17a2845c067c0413fdc5d7@172.30.0.10:5060
Contact: <sip:4169@172.30.0.186:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (14 headers 0 lines) ---
[2011-11-03 06:47:49] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1834367a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKc1452592C5F32C09
From: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (15 headers 0 lines) ---
[2011-11-03 06:47:49] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:49] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:49]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKc1452592C5F32C09;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as21c64c30
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4806050c"
Content-Length: 0


<------------>
[2011-11-03 06:47:49] Scheduling destruction of SIP dialog '76ce4e08-4f93254f-34c24d9e@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:49] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16274433
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36e753a2
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3e18f430bd400f654bff9128151289@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ec2b685
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as17f6477a
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f36c00a40aa5e3274b5842251035611@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #6 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK58f85e66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as21495120
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7cee79de70994b8d54e0bf3f175a74c4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK650f3ec4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1d2bed4e
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 092d23565b3c207e62cf74151550fd9b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3806872b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76b1d284
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60a65ea60b162e467a2d787f63ec3ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:49] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.192:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bKb1e2adab9542CE66
From: "4148" <sip:4148@customer.com>;tag=971A06C5-F38B6420
To: <sip:4148@customer.com>
CSeq: 1 REGISTER
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (12 headers 0 lines) ---
[2011-11-03 06:47:49] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:47:49]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bKb1e2adab9542CE66;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=971A06C5-F38B6420
To: <sip:4148@customer.com>;tag=as51f450d1
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.192
CSeq: 1 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a0346ab"
Content-Length: 0


<------------>
[2011-11-03 06:47:49] Scheduling destruction of SIP dialog '3b0be151-a656316c-99d39b7@172.30.0.192' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:49] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48329534
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as07c04e0f
To: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK605cfdb1
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
CSeq: 103 NOTIFY
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (11 headers 0 lines) ---
[2011-11-03 06:47:49] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:49] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3aa33491
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:49]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKa803a3dd6F0CEE3C
From: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="73a6a1e8", uri="sip:4122@customer.com:5060", response="2a170996cc84dc9d3f62d97f2fd56265", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:49] --- (16 headers 0 lines) ---
[2011-11-03 06:47:49] Creating new subscription
[2011-11-03 06:47:49] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:50] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:50] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog '5e4c9262-197fee59-47d4d9d8@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKa803a3dd6F0CEE3C;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as44540955
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:50] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:50] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:50] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Really destroying SIP dialog '4ed85f33-99e3da80-72a000c9@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK76ab7578
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as096c908b
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 35e9902d35cd8ea864f4dc4a709f3f6a@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4147' is now UNREACHABLE!  Last qualify: 0
[2011-11-03 06:47:50] Really destroying SIP dialog '35e9902d35cd8ea864f4dc4a709f3f6a@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d88f9ec
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3b3e3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2fc941da
To: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Really destroying SIP dialog 'f4c3d4c8-d2329751-4b92475a@172.30.0.180' Method: SUBSCRIBE
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4147[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKc33531dc87A7235
From: "Office 113" <sip:4171@customer.com>;tag=821408B2-3FC83533
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a0c62a6a-2b8a60b-569c1668@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:50] --- (15 headers 0 lines) ---
[2011-11-03 06:47:50] Creating new subscription
[2011-11-03 06:47:50] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:47:50] list_route: hop: <sip:4171@172.30.0.165:5060>
[2011-11-03 06:47:50] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKc33531dc87A7235;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=821408B2-3FC83533
To: <sip:4122@customer.com>;tag=as13e7e702
Call-ID: a0c62a6a-2b8a60b-569c1668@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7383bd56"
Content-Length: 0


<------------>
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog 'a0c62a6a-2b8a60b-569c1668@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a32cc84
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b1aef55
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 472d4757150639851e7cbcf93968e4b8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a2db5cf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1102b6dd
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 663e81e1654cded60e01fdf44e33daa1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b930f17
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6c549c
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3993488f7a3e144950e29ff625fd2e42@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00741eda
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2acd3794
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4bb193d547bf5a492e52b25b08ab30a4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK25f4486eDB6F3505
From: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4141", realm="asterisk", nonce="76c2b89e", uri="sip:4109@customer.com:5060", response="7e96f112edc261dd4afbd9a8bdef5d29", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:50] --- (16 headers 0 lines) ---
[2011-11-03 06:47:50] Creating new subscription
[2011-11-03 06:47:50] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:50] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:50] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog '34d99d55-c99f65f4-fe46e6db@172.30.0.181' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK25f4486eDB6F3505;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
To: <sip:4109@customer.com>;tag=as6e8b7bde
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:50] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:47:50] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:47:50] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK38283a41
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6e8b7bde
To: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKb4c5c4a9A2667890
From: "4168" <sip:4168@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4168", realm="asterisk", nonce="2a586252", uri="sip:4148@customer.com:5060", response="b23b39f9880c29a8e53c9e2f1e94acbd", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:50] --- (16 headers 0 lines) ---
[2011-11-03 06:47:50] Creating new subscription
[2011-11-03 06:47:50] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:50] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:47:50] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKb4c5c4a9A2667890;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as6885fa63
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.187
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="054b95a0"
Content-Length: 0


<------------>
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK626093b8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b7fc95f
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1e025d894d55f530573b086334348435@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Really destroying SIP dialog '1e025d894d55f530573b086334348435@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK08076f7c
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
CSeq: 104 NOTIFY
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:50] --- (11 headers 0 lines) ---
[2011-11-03 06:47:50] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:50] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK38283a41
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6e8b7bde
To: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ab3e09e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as04208b25
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78216a6342355e9c5111f1265af3f94c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Really destroying SIP dialog '2ff02f7b-7c7d8f8c-62981375@172.30.0.176' Method: SUBSCRIBE
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK9af195ed8C5184B2
From: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
To: <sip:4138@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="7d321885", uri="sip:4138@customer.com:5060", response="62d4dc3a59bf87464a68553fbdbe316b", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:50] --- (16 headers 0 lines) ---
[2011-11-03 06:47:50] Creating new subscription
[2011-11-03 06:47:50] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:50] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:50] Looking for 4138 in default-local (domain customer.com:5060)
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog 'd4966781-a4f38ab6-81a022c7@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK9af195ed8C5184B2;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
To: <sip:4138@customer.com>;tag=as72d45b6b
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4138@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:50] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:50] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:50] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e4d0fdd
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as72d45b6b
To: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKa0c62a6a2B8A60B
From: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:50] --- (15 headers 0 lines) ---
[2011-11-03 06:47:50] Creating new subscription
[2011-11-03 06:47:50] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:47:50] list_route: hop: <sip:4135@172.30.0.177:5060>
[2011-11-03 06:47:50] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKa0c62a6a2B8A60B;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
To: <sip:4108@customer.com>;tag=as0a17f176
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="732d40f2"
Content-Length: 0


<------------>
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog '569c1668-7cf1ab91-1b38956@172.30.0.177' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK38283a41
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6e8b7bde
To: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16274433
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36e753a2
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3e18f430bd400f654bff9128151289@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ec2b685
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as17f6477a
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f36c00a40aa5e3274b5842251035611@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3806872b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as76b1d284
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60a65ea60b162e467a2d787f63ec3ff2@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4125' is now UNREACHABLE!  Last qualify: 0
[2011-11-03 06:47:50] Really destroying SIP dialog '60a65ea60b162e467a2d787f63ec3ff2@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4131 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:47:50]   == Extension Changed 4125[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa0111d05999FDB0E
From: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:50] --- (15 headers 0 lines) ---
[2011-11-03 06:47:50] Creating new subscription
[2011-11-03 06:47:50] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:50] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:50] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa0111d05999FDB0E;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
To: <sip:4140@customer.com>;tag=as00e218e3
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02ae16f8"
Content-Length: 0


<------------>
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog 'a088c697-bbc01fa0-e89a2629@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e4d0fdd
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as72d45b6b
To: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK58f85e66
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as21495120
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7cee79de70994b8d54e0bf3f175a74c4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK650f3ec4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1d2bed4e
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 092d23565b3c207e62cf74151550fd9b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:50] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e5747d-a10f1dc-810b9183@172.30.0.191 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 19130ms with no response
[2011-11-03 06:47:50] Really destroying SIP dialog '15b9cd88-ef829ad-97c025f6@172.30.0.195' Method: REGISTER
[2011-11-03 06:47:50]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK7182916fEB9AD43E
From: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4125", realm="asterisk", nonce="3768806d", uri="sip:4127@customer.com:5060", response="1e56925635cd104873e3373ef67ea5f0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:50] --- (16 headers 0 lines) ---
[2011-11-03 06:47:50] Creating new subscription
[2011-11-03 06:47:50] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:50] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:50] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:50] Scheduling destruction of SIP dialog '8303abf8-652cc7a3-461beda2@172.30.0.168' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:50]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK7182916fEB9AD43E;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>;tag=as7283c46d
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:50] set_destination: Parsing <sip:4125@172.30.0.168:5060> for address/port to send to
[2011-11-03 06:47:50] set_destination: set destination to 172.30.0.168:5060
[2011-11-03 06:47:50] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK595e9039
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7283c46d
To: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e4d0fdd
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as72d45b6b
To: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:50] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission bfee6331-b25fc778-66739a8f@172.30.0.195 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6363ms with no response
[2011-11-03 06:47:51] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 330f082c-b8c37221-ae9a447a@172.30.0.195 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6357ms with no response
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.196:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKf0fe5b0f8FA72154
From: "4108" <sip:4108@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4108@customer.com>
CSeq: 2 REGISTER
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4108", realm="asterisk", nonce="0374a33d", uri="sip:customer.com:5060", response="71167bc64affe60007b92658ff17cb27", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (13 headers 0 lines) ---
[2011-11-03 06:47:51] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:51] NOTICE[9661]: chan_sip.c:25638 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2011-11-03 06:47:51] Really destroying SIP dialog '7cee79de70994b8d54e0bf3f175a74c4@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:51] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75874c71
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as739514e7
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7b5f652c2e2c7ab35d83933d0f0ffbe6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKf0fe5b0f8FA72154;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=9762B8C9-A46E1B76
To: <sip:4108@customer.com>;tag=as21cbc307
Call-ID: e7b8eeb5-a95f4612-4fb66da3@172.30.0.196
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4108@172.30.0.196:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:51 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog 'e7b8eeb5-a95f4612-4fb66da3@172.30.0.196' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK38283a41
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6e8b7bde
To: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48329534
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as07c04e0f
To: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #5 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK595e9039
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7283c46d
To: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3a32cc84
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1b1aef55
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 472d4757150639851e7cbcf93968e4b8@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51] Really destroying SIP dialog '472d4757150639851e7cbcf93968e4b8@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK89849113486E06E0
From: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
To: <sip:4140@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="6f5e92fb", uri="sip:4140@customer.com:5060", response="07b5f518781280593f34df42ad223a4e", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (16 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:51] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:51] Looking for 4140 in default-local (domain customer.com:5060)
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog '2645a66c-fc9dca45-6725fa2a@172.30.0.167' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK89849113486E06E0;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
To: <sip:4140@customer.com>;tag=as46091936
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4140@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:51] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:51] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:51] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65fbc5b4
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as46091936
To: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a2db5cf
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1102b6dd
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 663e81e1654cded60e01fdf44e33daa1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51] Really destroying SIP dialog '663e81e1654cded60e01fdf44e33daa1@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:51] Retransmitting #10 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4b930f17
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6c549c
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 3993488f7a3e144950e29ff625fd2e42@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51] Really destroying SIP dialog '3993488f7a3e144950e29ff625fd2e42@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK00741eda
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2acd3794
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4bb193d547bf5a492e52b25b08ab30a4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKa2667890CED7E9DB
From: "4125" <sip:4125@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 8b77eefa-eedd49d-2bf144b4@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (15 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:47:51] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bKa2667890CED7E9DB;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>;tag=as421f65eb
Call-ID: 8b77eefa-eedd49d-2bf144b4@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="231ddffd"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog '8b77eefa-eedd49d-2bf144b4@172.30.0.168' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.192:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK251567bc3586D07
From: "4148" <sip:4148@customer.com>;tag=F4CE8981-1D79849C
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e56f2582-f4496c2d-8cab4c08@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (15 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4148@172.30.0.192:5060>
[2011-11-03 06:47:51] Found peer '4148' for '4148' from 172.30.0.192:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK251567bc3586D07;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=F4CE8981-1D79849C
To: <sip:4108@customer.com>;tag=as02c69ca2
Call-ID: e56f2582-f4496c2d-8cab4c08@172.30.0.192
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ee59480"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog 'e56f2582-f4496c2d-8cab4c08@172.30.0.192' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK595e9039
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7283c46d
To: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65fbc5b4
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as46091936
To: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e4d0fdd
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as72d45b6b
To: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.188:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK3282930dF5F8024
From: "4149" <sip:4149@customer.com>;tag=40191473-99EF82FA
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a884d693-fa67221a-d2c9b479@172.30.0.188
Contact: <sip:4149@172.30.0.188:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (15 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.188:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4149@172.30.0.188:5060>
[2011-11-03 06:47:51] Found peer '4149' for '4149' from 172.30.0.188:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.188:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK3282930dF5F8024;received=172.30.0.188
From: "4149" <sip:4149@customer.com>;tag=40191473-99EF82FA
To: <sip:4109@customer.com>;tag=as2c8c385e
Call-ID: a884d693-fa67221a-d2c9b479@172.30.0.188
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12df35f0"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog 'a884d693-fa67221a-d2c9b479@172.30.0.188' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK2cd33c15DA11627E
From: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (15 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:51] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK2cd33c15DA11627E;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
To: <sip:4147@customer.com>;tag=as746a85fa
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e102873"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog '3a0ed75b-16468bc-154fc1f1@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ab3e09e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as04208b25
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78216a6342355e9c5111f1265af3f94c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51] Really destroying SIP dialog '973fd306-61221967-8836730c@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK8442b2e3C1452592
From: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (15 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4134@172.30.0.179:5060>
[2011-11-03 06:47:51] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK8442b2e3C1452592;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as397e6250
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="24078e2c"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog 'c5f32c09-76ce4e08-4f93254f@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65fbc5b4
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as46091936
To: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7fe241bb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.176:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK328f31a8673C3731
From: "4122" <sip:4122@customer.com>;tag=8ADDFE22-C406D99B
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 2ff02f7b-7c7d8f8c-62981375@172.30.0.176
Contact: <sip:4122@172.30.0.176:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4122", realm="asterisk", nonce="60dcb9a4", uri="sip:4122@customer.com:5060", response="bff9ede8717ccd9ad846c7001d8c1866", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (16 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.176:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4122@172.30.0.176:5060>
[2011-11-03 06:47:51] Found peer '4122' for '4122' from 172.30.0.176:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.176:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK328f31a8673C3731;received=172.30.0.176
From: "4122" <sip:4122@customer.com>;tag=8ADDFE22-C406D99B
To: <sip:4122@customer.com>;tag=as7ea692f1
Call-ID: 2ff02f7b-7c7d8f8c-62981375@172.30.0.176
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2229d22c"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog '2ff02f7b-7c7d8f8c-62981375@172.30.0.176' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51] Really destroying SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.178' Method: SUBSCRIBE
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16274433
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36e753a2
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3e18f430bd400f654bff9128151289@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ec2b685
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as17f6477a
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f36c00a40aa5e3274b5842251035611@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK7aecdd9222DB8323
From: "4108" <sip:4108@customer.com>;tag=1B63D766-80BD747
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 973fd306-61221967-8836730c@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4108", realm="asterisk", nonce="50f1db1f", uri="sip:4148@customer.com:5060", response="d7a01ad53292b8d8b9ea3ef26916c0b3", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (16 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:47:51] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK7aecdd9222DB8323;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=1B63D766-80BD747
To: <sip:4148@customer.com>;tag=as649c6cf7
Call-ID: 973fd306-61221967-8836730c@172.30.0.196
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ab9993c"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog '973fd306-61221967-8836730c@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK595e9039
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7283c46d
To: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #8 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4106@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK99def73a9AF5E9C7
From: "Office 131" <sip:4113@customer.com>;tag=DE280EAE-6341C34B
To: <sip:4106@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7f8bf698-7add0bfd-1a758a86@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (15 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:51] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK99def73a9AF5E9C7;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=DE280EAE-6341C34B
To: <sip:4106@customer.com>;tag=as76f95d78
Call-ID: 7f8bf698-7add0bfd-1a758a86@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1743dccb"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog '7f8bf698-7add0bfd-1a758a86@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK650f3ec4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1d2bed4e
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 092d23565b3c207e62cf74151550fd9b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK38283a41
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6e8b7bde
To: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK84fda8269853BA89
From: "Office 105" <sip:4131@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:51] --- (15 headers 0 lines) ---
[2011-11-03 06:47:51] Creating new subscription
[2011-11-03 06:47:51] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:51] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:47:51] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:47:51]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK84fda8269853BA89;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>;tag=as790626cd
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51037460"
Content-Length: 0


<------------>
[2011-11-03 06:47:51] Scheduling destruction of SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65fbc5b4
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as46091936
To: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:51] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f684b68
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as59c8844e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 542617da67c722dc46a6620a527febd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:51] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.182:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK7446be9093AC63D9
From: "Office 108" <sip:4128@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4128@customer.com>
CSeq: 2 REGISTER
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4128", realm="asterisk", nonce="556799c2", uri="sip:customer.com:5060", response="1cf1bc4c4c0cc6d54de065de2b4b69c7", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (13 headers 0 lines) ---
[2011-11-03 06:47:52] Sending to 172.30.0.182:5060 (no NAT)
[2011-11-03 06:47:52] NOTICE[9661]: chan_sip.c:25638 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2011-11-03 06:47:52] Really destroying SIP dialog '4bb193d547bf5a492e52b25b08ab30a4@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68dd9a7f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as323ac70c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15954a8b7552c83923de3d1076bfe4cf@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK7446be9093AC63D9;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=BE119CB1-13AF9B52
To: <sip:4128@customer.com>;tag=as7fe7f883
Call-ID: d595f645-e42497c6-8787fb9f@172.30.0.182
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4128@172.30.0.182:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:52 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'd595f645-e42497c6-8787fb9f@172.30.0.182' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:52] Really destroying SIP dialog 'db6f3505-b4cfe424-a9e0f78b@172.30.0.199' Method: SUBSCRIBE
[2011-11-03 06:47:52] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75874c71
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as739514e7
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7b5f652c2e2c7ab35d83933d0f0ffbe6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:52] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e4d0fdd
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as72d45b6b
To: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK6d5f5fd7BF46E004
From: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (15 headers 0 lines) ---
[2011-11-03 06:47:52] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:52] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK6d5f5fd7BF46E004;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>;tag=as159032dc
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="64d27540"
Content-Length: 0


<------------>
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'd2df29bd-7b13dd2-3f183dd3@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKbae9903b781D0A2A
From: "4116" <sip:4116@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="2ef1682c", uri="sip:4108@customer.com:5060", response="2a865bfe966c4f054f604a38f0501278", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (16 headers 0 lines) ---
[2011-11-03 06:47:52] Creating new subscription
[2011-11-03 06:47:52] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:52] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:52] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKbae9903b781D0A2A;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>;tag=as751466a5
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="295fcd7d"
Content-Length: 0


<------------>
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'db6f3505-b4cfe424-a9e0f78b@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52] Really destroying SIP dialog 'df1ce224-c5b03179-e464acb2@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:47:52] Really destroying SIP dialog 'ad1a82a4-d535aa1d-aa83522@172.30.0.167' Method: REGISTER
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bKdf9491cbF3731D5C
From: "4147" <sip:4147@customer.com>;tag=2A535BE0-132874A9
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (15 headers 0 lines) ---
[2011-11-03 06:47:52] Creating new subscription
[2011-11-03 06:47:52] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:52] list_route: hop: <sip:4147@172.30.0.183:5060>
[2011-11-03 06:47:52] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bKdf9491cbF3731D5C;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=2A535BE0-132874A9
To: <sip:4115@customer.com>;tag=as30268d77
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.183
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45da8d60"
Content-Length: 0


<------------>
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.183' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ab3e09e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as04208b25
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78216a6342355e9c5111f1265af3f94c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK7182916fEB9AD43E
From: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="5df1d7a9", uri="sip:4127@customer.com:5060", response="10b0bbf355753dd80cf7aecbac6aee27", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (16 headers 0 lines) ---
[2011-11-03 06:47:52] Creating new subscription
[2011-11-03 06:47:52] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:47:52] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:47:52] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'a02e6f59-4d240880-e0e88a0b@172.30.0.178' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK7182916fEB9AD43E;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>;tag=as1ce2bd26
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:52] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK595e9039
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7283c46d
To: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK5bfe143253F9CCF5
From: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4125", realm="asterisk", nonce="4da15ab2", uri="sip:4109@customer.com:5060", response="66cea9d019983d35231e739c35993849", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (16 headers 0 lines) ---
[2011-11-03 06:47:52] Creating new subscription
[2011-11-03 06:47:52] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:52] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:52] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog '78031bfd-6cc68e94-cf2353cf@172.30.0.168' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK5bfe143253F9CCF5;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>;tag=as20a415b7
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:52] set_destination: Parsing <sip:4125@172.30.0.168:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.168:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5628303c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a415b7
To: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19f6d50b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK320acc182444C743
From: "4168" <sip:4168@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: db6bcec2-2e8acd45-563892bc@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (15 headers 0 lines) ---
[2011-11-03 06:47:52] Creating new subscription
[2011-11-03 06:47:52] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:47:52] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:47:52] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK320acc182444C743;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>;tag=as7f1e5aab
Call-ID: db6bcec2-2e8acd45-563892bc@172.30.0.187
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31eea1d8"
Content-Length: 0


<------------>
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'db6bcec2-2e8acd45-563892bc@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK6a23fe0276151943
From: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="64d27540", uri="sip:4127@customer.com:5060", response="cb10c83bf3f778a4f255a5d16b7084e0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (16 headers 0 lines) ---
[2011-11-03 06:47:52] Creating new subscription
[2011-11-03 06:47:52] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:47:52] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:47:52] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'd2df29bd-7b13dd2-3f183dd3@172.30.0.165' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK6a23fe0276151943;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>;tag=as159032dc
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:52] set_destination: Parsing <sip:4171@172.30.0.165:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.165:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ad70ab7
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as159032dc
To: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16274433
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36e753a2
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7a3e18f430bd400f654bff9128151289@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:52] Really destroying SIP dialog '7a3e18f430bd400f654bff9128151289@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:52] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ec2b685
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as17f6477a
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1f36c00a40aa5e3274b5842251035611@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:52] Really destroying SIP dialog '1f36c00a40aa5e3274b5842251035611@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:52] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5628303c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a415b7
To: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.169:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.169:5060;branch=z9hG4bKb9c00d0c3A0EBCA5
From: "4127" <sip:4127@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4127@customer.com>
CSeq: 2 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.169
Contact: <sip:4127@172.30.0.169:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4127", realm="asterisk", nonce="45ab85f9", uri="sip:customer.com:5060", response="58775eddcf59652d5581649ab1ade755", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (13 headers 0 lines) ---
[2011-11-03 06:47:52] Sending to 172.30.0.169:5060 (no NAT)
[2011-11-03 06:47:52]     -- Registered SIP '4127' at 172.30.0.169:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3a6fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as42b0f452
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6514502c37c16ff455cdada32716a07b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.169:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.169:5060;branch=z9hG4bKb9c00d0c3A0EBCA5;received=172.30.0.169
From: "4127" <sip:4127@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4127@customer.com>;tag=as3a50f2aa
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.169
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.169:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:52 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.169' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:52] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65fbc5b4
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as46091936
To: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:52] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK650f3ec4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1d2bed4e
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 092d23565b3c207e62cf74151550fd9b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4116 (queued)
[2011-11-03 06:47:52] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4120
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4128 (queued)
[2011-11-03 06:47:52] set_destination: Parsing <sip:4125@172.30.0.168:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.168:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d85cd6c
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as0a1f07c6
To: "4125" <sip:4125@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4125
[2011-11-03 06:47:52] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47a91343
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4136
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:52] set_destination: Parsing <sip:4138@172.30.0.163:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.163:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c484883
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as127a9232
To: "4138" <sip:4138@customer.com>;tag=C47CD70A-391C653F
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6351d728-8b1f2125-2caa38e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4138
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:47:52] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b31f9f3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4135
[2011-11-03 06:47:52] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4886abd9
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4145
[2011-11-03 06:47:52] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK739c3275
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4113
[2011-11-03 06:47:52] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:52] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:52] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK71fd35c3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4118
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4125 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:47:52]   == Extension Changed 4127[default-local] new state Idle for Notify User 4171 (queued)
[2011-11-03 06:47:52]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK8442b2e3C1452592
From: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:52] --- (15 headers 0 lines) ---
[2011-11-03 06:47:52] Creating new subscription
[2011-11-03 06:47:52] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:52] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:47:52] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:52]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK8442b2e3C1452592;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as24d6fe6e
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7ceefd6a"
Content-Length: 0


<------------>
[2011-11-03 06:47:52] Scheduling destruction of SIP dialog 'c5f32c09-76ce4e08-4f93254f@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:52] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ad70ab7
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as159032dc
To: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5628303c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a415b7
To: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d85cd6c
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as0a1f07c6
To: "4125" <sip:4125@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c484883
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as127a9232
To: "4138" <sip:4138@customer.com>;tag=C47CD70A-391C653F
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6351d728-8b1f2125-2caa38e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:52] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b31f9f3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK7182916fEB9AD43E
From: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="5df1d7a9", uri="sip:4127@customer.com:5060", response="10b0bbf355753dd80cf7aecbac6aee27", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (16 headers 0 lines) ---
[2011-11-03 06:47:53] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:53] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog 'a02e6f59-4d240880-e0e88a0b@172.30.0.178' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK7182916fEB9AD43E;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
To: <sip:4127@customer.com>;tag=as1ce2bd26
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:53] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:47:53] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:47:53] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4886abd9
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK739c3275
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Really destroying SIP dialog '198759bc-5af6fe25-2cc894a6@172.30.0.182' Method: SUBSCRIBE
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f684b68
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as59c8844e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 542617da67c722dc46a6620a527febd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ad70ab7
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as159032dc
To: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK71fd35c3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK8d39ca246589A73
From: "4140" <sip:4140@customer.com>;tag=1B63D766-80BD747
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ebf68d48-834d5031-6928867e@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (15 headers 0 lines) ---
[2011-11-03 06:47:53] Creating new subscription
[2011-11-03 06:47:53] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:47:53] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:47:53] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK8d39ca246589A73;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=1B63D766-80BD747
To: <sip:4122@customer.com>;tag=as74d18b5a
Call-ID: ebf68d48-834d5031-6928867e@172.30.0.175
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d19bee0"
Content-Length: 0


<------------>
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog 'ebf68d48-834d5031-6928867e@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75874c71
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as739514e7
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7b5f652c2e2c7ab35d83933d0f0ffbe6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d85cd6c
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as0a1f07c6
To: "4125" <sip:4125@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c484883
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as127a9232
To: "4138" <sip:4138@customer.com>;tag=C47CD70A-391C653F
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6351d728-8b1f2125-2caa38e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b31f9f3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4886abd9
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68dd9a7f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as323ac70c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15954a8b7552c83923de3d1076bfe4cf@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK3e0d6f69B3C8FD62
From: "Office 131" <sip:4113@customer.com>;tag=AF4DF18F-59762C00
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 1da9150f-2b3ef580-8d99cf05@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (15 headers 0 lines) ---
[2011-11-03 06:47:53] Creating new subscription
[2011-11-03 06:47:53] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:53] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:53] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK3e0d6f69B3C8FD62;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=AF4DF18F-59762C00
To: <sip:4138@customer.com>;tag=as59a64c02
Call-ID: 1da9150f-2b3ef580-8d99cf05@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23b3c9a9"
Content-Length: 0


<------------>
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog '1da9150f-2b3ef580-8d99cf05@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK739c3275
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #12 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK7dd1876d60DE438C
From: "Office 105" <sip:4131@customer.com>;tag=7AFFE505-8E3E0084
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: b86e3dc7-5211810e-25a67e11@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (15 headers 0 lines) ---
[2011-11-03 06:47:53] Creating new subscription
[2011-11-03 06:47:53] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:53] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:47:53] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK7dd1876d60DE438C;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=7AFFE505-8E3E0084
To: <sip:4108@customer.com>;tag=as04493bcd
Call-ID: b86e3dc7-5211810e-25a67e11@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02668970"
Content-Length: 0


<------------>
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog 'b86e3dc7-5211810e-25a67e11@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5628303c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a415b7
To: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1202cbc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e261fea
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5de5c66b52b6d3c7210b2200441fbaa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47a91343
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK71fd35c3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ad70ab7
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as159032dc
To: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKd2df29bd7B13DD2
From: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (15 headers 0 lines) ---
[2011-11-03 06:47:53] Creating new subscription
[2011-11-03 06:47:53] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:47:53] list_route: hop: <sip:4135@172.30.0.177:5060>
[2011-11-03 06:47:53] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKd2df29bd7B13DD2;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
To: <sip:4122@customer.com>;tag=as2f99879a
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61ab4183"
Content-Length: 0


<------------>
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog '3f183dd3-50755910-9d8e4599@172.30.0.177' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:53] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK38283a41
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6e8b7bde
To: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ab3e09e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as04208b25
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 78216a6342355e9c5111f1265af3f94c@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:53] Really destroying SIP dialog '78216a6342355e9c5111f1265af3f94c@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:53] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48329534
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as07c04e0f
To: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d85cd6c
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as0a1f07c6
To: "4125" <sip:4125@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c484883
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as127a9232
To: "4138" <sip:4138@customer.com>;tag=C47CD70A-391C653F
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6351d728-8b1f2125-2caa38e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34c24d9eBDF5D4B5
From: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4134", realm="asterisk", nonce="0ba5e4a7", uri="sip:4115@customer.com:5060", response="44061039bf35e9c494289f1fce66a6ec", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (16 headers 0 lines) ---
[2011-11-03 06:47:53] Creating new subscription
[2011-11-03 06:47:53] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:53] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:53] Looking for 4115 in default-local (domain customer.com:5060)
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.179' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34c24d9eBDF5D4B5;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as771d3887
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4115@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:53] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:47:53] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:47:53] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78841311
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b31f9f3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4886abd9
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK739c3275
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1834367a
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #7 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65e2f80e
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as260c65d7
To: "empty station" <sip:4119@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: f7b8b3e-34d99d55-c99f65f4@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKd7fdcde12843320
From: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="4806050c", uri="sip:4147@customer.com:5060", response="682c104caba32ce6c1b6aa2b0f1a4873", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (16 headers 0 lines) ---
[2011-11-03 06:47:53] Creating new subscription
[2011-11-03 06:47:53] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:53] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:53] Looking for 4147 in default-local (domain customer.com:5060)
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog '76ce4e08-4f93254f-34c24d9e@172.30.0.199' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKd7fdcde12843320;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as21c64c30
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4147@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:53] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:47:53] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:47:53] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06f91a0a
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as21c64c30
To: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Really destroying SIP dialog 'ddf9dab8-ab9de2a1-75ab5826@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:47:53] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7e4d0fdd
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as72d45b6b
To: "4145" <sip:4145@customer.com>;tag=A37B38FD-E92D7A82
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: d4966781-a4f38ab6-81a022c7@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK650f3ec4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1d2bed4e
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 092d23565b3c207e62cf74151550fd9b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:53] Really destroying SIP dialog '092d23565b3c207e62cf74151550fd9b@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:53] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Retransmitting #1 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3a6fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as42b0f452
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6514502c37c16ff455cdada32716a07b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:53] Retransmitting #5 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3aa33491
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:53] Really destroying SIP dialog '8303abf8-652cc7a3-461beda2@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:47:53]
<--- SIP read from UDP:172.30.0.192:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK2c9800135ECF34E
From: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:53] --- (15 headers 0 lines) ---
[2011-11-03 06:47:53] Creating new subscription
[2011-11-03 06:47:53] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:47:53] list_route: hop: <sip:4148@172.30.0.192:5060>
[2011-11-03 06:47:53] Found peer '4148' for '4148' from 172.30.0.192:5060
[2011-11-03 06:47:53]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK2c9800135ECF34E;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
To: <sip:4115@customer.com>;tag=as3f9e7318
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="695fb4bc"
Content-Length: 0


<------------>
[2011-11-03 06:47:53] Scheduling destruction of SIP dialog '312cc0b9-ab7a3354-7ac3461f@172.30.0.192' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4136@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKe00528da83EB8A1B
From: "4145" <sip:4145@customer.com>;tag=4B97035A-1372D29B
To: <sip:4136@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: e8fa9348-b82111f1-f5635466@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (15 headers 0 lines) ---
[2011-11-03 06:47:54] Creating new subscription
[2011-11-03 06:47:54] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:54] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:54] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKe00528da83EB8A1B;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=4B97035A-1372D29B
To: <sip:4136@customer.com>;tag=as1601d4dc
Call-ID: e8fa9348-b82111f1-f5635466@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2170ec07"
Content-Length: 0


<------------>
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog 'e8fa9348-b82111f1-f5635466@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06f91a0a
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as21c64c30
To: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f684b68
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as59c8844e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 542617da67c722dc46a6620a527febd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2b192e34898B92BD
From: "Office 106" <sip:4120@customer.com>;tag=1296CD8-F3BE4761
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f4c3d4c8-d2329751-4b92475a@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="54839435", uri="sip:4115@customer.com:5060", response="cb757fee90b5d91ef8183762d6d1b3c2", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (16 headers 0 lines) ---
[2011-11-03 06:47:54] Creating new subscription
[2011-11-03 06:47:54] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:54] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:54] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2b192e34898B92BD;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=1296CD8-F3BE4761
To: <sip:4115@customer.com>;tag=as022a49ca
Call-ID: f4c3d4c8-d2329751-4b92475a@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e788e5b"
Content-Length: 0


<------------>
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog 'f4c3d4c8-d2329751-4b92475a@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK71fd35c3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d88f9ec
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3b3e3
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as2fc941da
To: "4118" <sip:4118@customer.com>;tag=BF73B7-F91F4BEE
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: b3f25e9-a004f0c0-87332547@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5628303c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a415b7
To: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Really destroying SIP dialog '7fd09539-8cc4f29a-6f06e753@172.30.0.176' Method: SUBSCRIBE
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.184:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKae24e90e15D5ECAF
From: "4123" <sip:4123@customer.com>;tag=2CB6E1E9-6C3CD0A6
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 4f4a86e4-c1c0933d-d5d4f48a@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (15 headers 0 lines) ---
[2011-11-03 06:47:54] Creating new subscription
[2011-11-03 06:47:54] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:54] list_route: hop: <sip:4123@172.30.0.184:5060>
[2011-11-03 06:47:54] Found peer '4123' for '4123' from 172.30.0.184:5060
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKae24e90e15D5ECAF;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=2CB6E1E9-6C3CD0A6
To: <sip:4148@customer.com>;tag=as384232f4
Call-ID: 4f4a86e4-c1c0933d-d5d4f48a@172.30.0.184
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42bb52a3"
Content-Length: 0


<------------>
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog '4f4a86e4-c1c0933d-d5d4f48a@172.30.0.184' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54] Retransmitting #5 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK595e9039
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7283c46d
To: "4125" <sip:4125@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 8303abf8-652cc7a3-461beda2@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75874c71
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as739514e7
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7b5f652c2e2c7ab35d83933d0f0ffbe6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06674834
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b3df67b
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7555d65d29f62798040af4023af79a54@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68dd9a7f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as323ac70c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15954a8b7552c83923de3d1076bfe4cf@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06f91a0a
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as21c64c30
To: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ad70ab7
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as159032dc
To: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34c24d9eBDF5D4B5
From: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4134", realm="asterisk", nonce="0ba5e4a7", uri="sip:4115@customer.com:5060", response="44061039bf35e9c494289f1fce66a6ec", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (16 headers 0 lines) ---
[2011-11-03 06:47:54] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:54] Looking for 4115 in default-local (domain customer.com:5060)
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.179' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK34c24d9eBDF5D4B5;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as771d3887
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4115@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:54] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:47:54] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:47:54] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25d27825
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Really destroying SIP dialog 'f064cb5f-1e17ff3c-d80a02d5@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:47:54] Retransmitting #6 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Really destroying SIP dialog 'c60da669-7b8a5940-3e53f5c7@172.30.0.173' Method: REGISTER
[2011-11-03 06:47:54] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65fbc5b4
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as46091936
To: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa0111d05999FDB0E
From: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (15 headers 0 lines) ---
[2011-11-03 06:47:54] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:54] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa0111d05999FDB0E;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
To: <sip:4140@customer.com>;tag=as00e218e3
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02ae16f8"
Content-Length: 0


<------------>
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog 'a088c697-bbc01fa0-e89a2629@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission f89e3598-d42ca483-7eea913a@172.30.0.171 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6447ms with no response
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d85cd6c
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as0a1f07c6
To: "4125" <sip:4125@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47a91343
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c484883
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as127a9232
To: "4138" <sip:4138@customer.com>;tag=C47CD70A-391C653F
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6351d728-8b1f2125-2caa38e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b31f9f3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4886abd9
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK739c3275
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKfe46e6db1C0E3CA
From: "4116" <sip:4116@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 43c45281-77964ac0-3ae67647@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (15 headers 0 lines) ---
[2011-11-03 06:47:54] Creating new subscription
[2011-11-03 06:47:54] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:54] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:54] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKfe46e6db1C0E3CA;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as3b315700
Call-ID: 43c45281-77964ac0-3ae67647@172.30.0.199
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b8e9866"
Content-Length: 0


<------------>
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog '43c45281-77964ac0-3ae67647@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54] Really destroying SIP dialog '16a0ba78-aedee661-2e9fbe7e@172.30.0.184' Method: SUBSCRIBE
[2011-11-03 06:47:54] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1202cbc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e261fea
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5de5c66b52b6d3c7210b2200441fbaa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKd5bc88df2AC04DE8
From: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
To: <sip:4119@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="1aa894ea", uri="sip:4119@customer.com:5060", response="b50ce21aeb7cf153c69bd995143119a7", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (16 headers 0 lines) ---
[2011-11-03 06:47:54] Creating new subscription
[2011-11-03 06:47:54] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:54] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:54] Looking for 4119 in default-local (domain customer.com:5060)
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog '136b5158-e366ebe1-5d9773ea@172.30.0.180' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKd5bc88df2AC04DE8;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
To: <sip:4119@customer.com>;tag=as773db5d8
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4119@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:54] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:47:54] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:47:54] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK324b0da5
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as773db5d8
To: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Really destroying SIP dialog 'e56f2582-f4496c2d-8cab4c08@172.30.0.166' Method: SUBSCRIBE
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK34c24d9eBDF5D4B5
From: "4124" <sip:4124@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 60ff2a54-bae9903b-781d0a2a@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (15 headers 0 lines) ---
[2011-11-03 06:47:54] Creating new subscription
[2011-11-03 06:47:54] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:54] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:47:54] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK34c24d9eBDF5D4B5;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as2f78562a
Call-ID: 60ff2a54-bae9903b-781d0a2a@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ab46a55"
Content-Length: 0


<------------>
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog '60ff2a54-bae9903b-781d0a2a@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06f91a0a
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as21c64c30
To: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Really destroying SIP dialog 'e42bb03b-4ddb3960-32efff69@172.30.0.184' Method: SUBSCRIBE
[2011-11-03 06:47:54] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b51d386
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as094a02b4
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4946cd7224ccf4901ee58df76f84ffee@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK9708841f46781D2C
From: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:54] --- (15 headers 0 lines) ---
[2011-11-03 06:47:54] Creating new subscription
[2011-11-03 06:47:54] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:47:54] list_route: hop: <sip:4135@172.30.0.177:5060>
[2011-11-03 06:47:54] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:47:54]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK9708841f46781D2C;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as74380269
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01949029"
Content-Length: 0


<------------>
[2011-11-03 06:47:54] Scheduling destruction of SIP dialog '65ddf445-1153d83a-5a70009b@172.30.0.177' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:54] Reliably Transmitting (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK67d60ed2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as77df2ea8
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62ad788e2212ca57369e7f2b53994167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK324b0da5
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as773db5d8
To: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:54] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1206b3ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4ee4835b
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 526bd48525ebf28d60c7c699612083fb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54] Really destroying SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.199' Method: SUBSCRIBE
[2011-11-03 06:47:54] Retransmitting #2 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3a6fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as42b0f452
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6514502c37c16ff455cdada32716a07b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:54] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.184:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKcddfff3fD329ECB4
From: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4123", realm="asterisk", nonce="53d536ce", uri="sip:4109@customer.com:5060", response="c888d1e2bd751d7f652982794c01e746", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (16 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:47:55] Found peer '4123' for '4123' from 172.30.0.184:5060
[2011-11-03 06:47:55] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog '6b074147-5962c75c-3c206d15@172.30.0.184' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKcddfff3fD329ECB4;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
To: <sip:4109@customer.com>;tag=as1bf9566b
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:55] set_destination: Parsing <sip:4123@172.30.0.184:5060> for address/port to send to
[2011-11-03 06:47:55] set_destination: set destination to 172.30.0.184:5060
[2011-11-03 06:47:55] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcbd0a0
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as1bf9566b
To: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Really destroying SIP dialog '85339b5d-941df874-9f68d33b@172.30.0.173' Method: SUBSCRIBE
[2011-11-03 06:47:55] Retransmitting #4 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f684b68
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as59c8844e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 542617da67c722dc46a6620a527febd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK324b0da5
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as773db5d8
To: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK29412cbf70932DC4
From: "4140" <sip:4140@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 32de0b9d-c79fec9a-7fe39c4b@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (15 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:47:55] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:47:55] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK29412cbf70932DC4;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4108@customer.com>;tag=as22dd38f6
Call-ID: 32de0b9d-c79fec9a-7fe39c4b@172.30.0.175
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61ba09ec"
Content-Length: 0


<------------>
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog '32de0b9d-c79fec9a-7fe39c4b@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK75874c71
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as739514e7
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7b5f652c2e2c7ab35d83933d0f0ffbe6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55] Really destroying SIP dialog '7b5f652c2e2c7ab35d83933d0f0ffbe6@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKd7fdcde12843320
From: "4124" <sip:4124@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a76923a7-aea30b36-e90f478d@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (15 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:47:55] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:47:55] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bKd7fdcde12843320;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>;tag=as64914f5d
Call-ID: a76923a7-aea30b36-e90f478d@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12b90c0a"
Content-Length: 0


<------------>
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog 'a76923a7-aea30b36-e90f478d@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55] Really destroying SIP dialog 'c60da669-7b8a5940-3e53f5c7@172.30.0.185' Method: REGISTER
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcbd0a0
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as1bf9566b
To: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #11 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68dd9a7f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as323ac70c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15954a8b7552c83923de3d1076bfe4cf@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06674834
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b3df67b
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7555d65d29f62798040af4023af79a54@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78841311
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKa523c1cdB4D24F8A
From: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4108", realm="asterisk", nonce="08488643", uri="sip:4109@customer.com:5060", response="91b4d1d074b2cb74aced8090e1d194a6", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (16 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:55] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:47:55] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog '834d5031-6928867e-29412cbf@172.30.0.196' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bKa523c1cdB4D24F8A;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4109@customer.com>;tag=as53f2e80f
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:55] set_destination: Parsing <sip:4108@172.30.0.196:5060> for address/port to send to
[2011-11-03 06:47:55] set_destination: set destination to 172.30.0.196:5060
[2011-11-03 06:47:55] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4783d2c2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as53f2e80f
To: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcbd0a0
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as1bf9566b
To: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06f91a0a
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as21c64c30
To: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK71fd35c3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK2ff02f7b7C7D8F8C
From: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4147", realm="asterisk", nonce="4c88c375", uri="sip:4148@customer.com:5060", response="7bf30c7b45dd2f7b73b681501304a714", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (16 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:47:55] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:47:55] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog '198759bc-5af6fe25-2cc894a6@172.30.0.183' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bK2ff02f7b7C7D8F8C;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
To: <sip:4148@customer.com>;tag=as32698f53
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:55] set_destination: Parsing <sip:4147@172.30.0.183:5060> for address/port to send to
[2011-11-03 06:47:55] set_destination: set destination to 172.30.0.183:5060
[2011-11-03 06:47:55] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d4d4415
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as32698f53
To: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1202cbc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e261fea
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5de5c66b52b6d3c7210b2200441fbaa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4136@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK771690b07E0815F5
From: "Office 131" <sip:4113@customer.com>;tag=3C349D18-BE1E547D
To: <sip:4136@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 13d339de-a62eda3b-c4eedd1c@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (15 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:55] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:55] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK771690b07E0815F5;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=3C349D18-BE1E547D
To: <sip:4136@customer.com>;tag=as54880bed
Call-ID: 13d339de-a62eda3b-c4eedd1c@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01cd93c0"
Content-Length: 0


<------------>
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog '13d339de-a62eda3b-c4eedd1c@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4783d2c2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as53f2e80f
To: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b3f25e9-a004f0c0-87332547@172.30.0.173 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10663ms with no response
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.192:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK831c241aC84A2845
From: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4148", realm="asterisk", nonce="1bdc9853", uri="sip:4122@customer.com:5060", response="2d194b2d2edbe73edb225bad3cf275f2", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (16 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:47:55] Found peer '4148' for '4148' from 172.30.0.192:5060
[2011-11-03 06:47:55] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog 'cccbccfb-9f3dfab6-424f6aa1@172.30.0.192' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK831c241aC84A2845;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>;tag=as252ea80a
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:55] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:47:55] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:47:55] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9261d0
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as252ea80a
To: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7fe241bb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d4d4415
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as32698f53
To: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK10a45f63
From: <sip:4122@customer.com>;tag=as539e8ced
To: "4106" <sip:4106@customer.com>;tag=79DAA10-211CA7DB
CSeq: 102 NOTIFY
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (11 headers 0 lines) ---
[2011-11-03 06:47:55] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK324b0da5
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as773db5d8
To: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #5 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5628303c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a415b7
To: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4783d2c2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as53f2e80f
To: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK60ff2a54BAE9903B
From: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="42c0d266", uri="sip:4108@customer.com:5060", response="bc06628508bee276acf427c49952a0ee", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (16 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:47:55] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:47:55] Looking for 4108 in default-local (domain customer.com:5060)
[2011-11-03 06:47:55] Scheduling destruction of SIP dialog 'db6f3505-b4cfe424-a9e0f78b@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:55]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK60ff2a54BAE9903B;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
To: <sip:4108@customer.com>;tag=as3a063028
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4108@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:55] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:55] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:55] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07707bac
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as3a063028
To: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #9 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcbd0a0
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as1bf9566b
To: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1834367a
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 102 NOTIFY
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (11 headers 0 lines) ---
[2011-11-03 06:47:55] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:55] Retransmitting #3 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3a6fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as42b0f452
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6514502c37c16ff455cdada32716a07b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d4d4415
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as32698f53
To: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #5 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ad70ab7
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as159032dc
To: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Really destroying SIP dialog 'b7988ba2-27b08275-7d13db4@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b51d386
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as094a02b4
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4946cd7224ccf4901ee58df76f84ffee@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55] Really destroying SIP dialog 'f190f9ad-ecdfb3b6-25029b3f@172.30.0.180' Method: REGISTER
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK16274433
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as36e753a2
To: "4123" <sip:4123@172.30.0.184:5060>;tag=CA5517DA-DDCEEB6B
CSeq: 102 OPTIONS
Call-ID: 7a3e18f430bd400f654bff9128151289@172.30.0.10:5060
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (14 headers 0 lines) ---
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25d27825
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK67d60ed2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as77df2ea8
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62ad788e2212ca57369e7f2b53994167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55] Retransmitting #5 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d85cd6c
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as0a1f07c6
To: "4125" <sip:4125@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.168
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:55] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1206b3ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4ee4835b
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 526bd48525ebf28d60c7c699612083fb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:55]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK70932dc432DE0B9D
From: "4108" <sip:4108@customer.com>;tag=AF8644A8-D23ACD11
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c79fec9a-7fe39c4b-fb985080@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:55] --- (15 headers 0 lines) ---
[2011-11-03 06:47:55] Creating new subscription
[2011-11-03 06:47:55] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:55] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:47:56] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:47:56]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK70932dc432DE0B9D;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=AF8644A8-D23ACD11
To: <sip:4127@customer.com>;tag=as66c006e7
Call-ID: c79fec9a-7fe39c4b-fb985080@172.30.0.196
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09ebc6b5"
Content-Length: 0


<------------>
[2011-11-03 06:47:56] Scheduling destruction of SIP dialog 'c79fec9a-7fe39c4b-fb985080@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:56] Retransmitting #5 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c484883
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as127a9232
To: "4138" <sip:4138@customer.com>;tag=C47CD70A-391C653F
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6351d728-8b1f2125-2caa38e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #5 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b31f9f3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4886abd9
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK739c3275
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5f684b68
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as59c8844e
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 542617da67c722dc46a6620a527febd6@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:56] Really destroying SIP dialog '542617da67c722dc46a6620a527febd6@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:56]
<--- SIP read from UDP:172.30.0.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3aa33491
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
CSeq: 102 NOTIFY
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


------------->
[2011-11-03 06:47:56] --- (11 headers 0 lines) ---
[2011-11-03 06:47:56] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:56] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07707bac
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as3a063028
To: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4783d2c2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as53f2e80f
To: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9261d0
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as252ea80a
To: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKd50400c2120071B9
From: "4116" <sip:4116@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="6dba1b29", uri="sip:4115@customer.com:5060", response="b3933f632816f57f0e6078b925cee770", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:56] --- (16 headers 0 lines) ---
[2011-11-03 06:47:56] Creating new subscription
[2011-11-03 06:47:56] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:47:56] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:47:56] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:56]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKd50400c2120071B9;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as4f0abc0f
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fbcbf17"
Content-Length: 0


<------------>
[2011-11-03 06:47:56] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK68dd9a7f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as323ac70c
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 15954a8b7552c83923de3d1076bfe4cf@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:56] Really destroying SIP dialog '15954a8b7552c83923de3d1076bfe4cf@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:56] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d4d4415
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as32698f53
To: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKd50cf517379C317C
From: "4140" <sip:4140@customer.com>;tag=AF8644A8-D23ACD11
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 661f2035-7aecdd92-22db8323@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:56] --- (15 headers 0 lines) ---
[2011-11-03 06:47:56] Creating new subscription
[2011-11-03 06:47:56] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:47:56] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:47:56] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:47:56]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKd50cf517379C317C;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=AF8644A8-D23ACD11
To: <sip:4115@customer.com>;tag=as6c4b8054
Call-ID: 661f2035-7aecdd92-22db8323@172.30.0.175
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f3d3cd6"
Content-Length: 0


<------------>
[2011-11-03 06:47:56] Scheduling destruction of SIP dialog '661f2035-7aecdd92-22db8323@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:56] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06674834
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b3df67b
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7555d65d29f62798040af4023af79a54@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:56] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission f7b8b3e-34d99d55-c99f65f4@172.30.0.191 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 19285ms with no response
[2011-11-03 06:47:56] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47a91343
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1202cbc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e261fea
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5de5c66b52b6d3c7210b2200441fbaa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:56] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4135@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK424c20395C47284
From: "Office 131" <sip:4113@customer.com>;tag=30F00A61-FAEDD1BA
To: <sip:4135@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7f0aa159-11595212-dcaaf37f@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:56] --- (15 headers 0 lines) ---
[2011-11-03 06:47:56] Creating new subscription
[2011-11-03 06:47:56] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:56] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:47:56] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:56]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK424c20395C47284;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=30F00A61-FAEDD1BA
To: <sip:4135@customer.com>;tag=as6a25eebf
Call-ID: 7f0aa159-11595212-dcaaf37f@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50a04d96"
Content-Length: 0


<------------>
[2011-11-03 06:47:56] Scheduling destruction of SIP dialog '7f0aa159-11595212-dcaaf37f@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19f6d50b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcbd0a0
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as1bf9566b
To: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 34d99d55-c99f65f4-fe46e6db@172.30.0.181 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6341ms with no response
[2011-11-03 06:47:56] Really destroying SIP dialog 'a0c62a6a-2b8a60b-569c1668@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:47:56]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK60ff2a54BAE9903B
From: "Leon Zhong" <sip:4134@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 781d0a2a-d7fdcde1-2843320@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:56] --- (15 headers 0 lines) ---
[2011-11-03 06:47:56] Creating new subscription
[2011-11-03 06:47:56] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:56] list_route: hop: <sip:4134@172.30.0.179:5060>
[2011-11-03 06:47:56] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:56]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK60ff2a54BAE9903B;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>;tag=as2e21d9a2
Call-ID: 781d0a2a-d7fdcde1-2843320@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fa691ee"
Content-Length: 0


<------------>
[2011-11-03 06:47:56] Scheduling destruction of SIP dialog '781d0a2a-d7fdcde1-2843320@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:56] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07707bac
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as3a063028
To: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK38283a41
From: <sip:4109@customer.com>;tag=as6e8b7bde
To: "Office 107" <sip:4141@customer.com>;tag=EEC06C12-96421E89
CSeq: 102 NOTIFY
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:56] --- (11 headers 0 lines) ---
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3a6fe
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as42b0f452
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6514502c37c16ff455cdada32716a07b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:56] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4127' is now UNREACHABLE!  Last qualify: 0
[2011-11-03 06:47:56] Really destroying SIP dialog '6514502c37c16ff455cdada32716a07b@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:56] Really destroying SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:47:56] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b51d386
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as094a02b4
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4946cd7224ccf4901ee58df76f84ffee@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:56] Really destroying SIP dialog '52853a3a-7c6d73e1-2ee1b038@172.30.0.173' Method: SUBSCRIBE
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK324b0da5
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as773db5d8
To: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission d4966781-a4f38ab6-81a022c7@172.30.0.167 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6313ms with no response
[2011-11-03 06:47:56] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4783d2c2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as53f2e80f
To: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4116 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4128 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4125 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4135 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4125 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:47:56]   == Extension Changed 4127[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKaacb1a329C673BBB
From: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (15 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:47:57] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKaacb1a329C673BBB;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
To: <sip:4138@customer.com>;tag=as0e9574c5
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="218e7b27"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog 'fde2cac4-4612074d-b2293156@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] Really destroying SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.191' Method: REGISTER
[2011-11-03 06:47:57] Retransmitting #2 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK67d60ed2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as77df2ea8
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62ad788e2212ca57369e7f2b53994167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:57] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1206b3ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4ee4835b
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 526bd48525ebf28d60c7c699612083fb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:57] Retransmitting #5 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06f91a0a
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as21c64c30
To: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d4d4415
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as32698f53
To: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.176:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK5069c9a4364A15CD
From: "4122" <sip:4122@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.176
Contact: <sip:4122@172.30.0.176:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (15 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.176:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4122@172.30.0.176:5060>
[2011-11-03 06:47:57] Found peer '4122' for '4122' from 172.30.0.176:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.176:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.176:5060;branch=z9hG4bK5069c9a4364A15CD;received=172.30.0.176
From: "4122" <sip:4122@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>;tag=as16e221f9
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.176
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78479f30"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog 'cfe67e0e-c5367aa7-166259d8@172.30.0.176' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9261d0
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as252ea80a
To: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK29be0d3b470771D2
From: "Office 105" <sip:4131@customer.com>;tag=4FDF6BD7-7F3C8D5E
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 1844f365-7b9fec64-c3d42aff@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (15 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:47:57] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK29be0d3b470771D2;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=4FDF6BD7-7F3C8D5E
To: <sip:4127@customer.com>;tag=as531f15e9
Call-ID: 1844f365-7b9fec64-c3d42aff@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="097c6a15"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog '1844f365-7b9fec64-c3d42aff@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06674834
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b3df67b
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7555d65d29f62798040af4023af79a54@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:57] Retransmitting #13 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKcbcf9172A3BA2D85
From: "4115" <sip:4115@customer.com>;tag=E84B1746-E4C25229
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: b7988ba2-27b08275-7d13db4@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="52e44a83", uri="sip:4115@customer.com:5060", response="3cbb09316f82f34a0cfc2ec6f5965597", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (16 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4115@172.30.0.197:5060>
[2011-11-03 06:47:57] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKcbcf9172A3BA2D85;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=E84B1746-E4C25229
To: <sip:4115@customer.com>;tag=as68ee3032
Call-ID: b7988ba2-27b08275-7d13db4@172.30.0.197
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b3e0324"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog 'b7988ba2-27b08275-7d13db4@172.30.0.197' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 8303abf8-652cc7a3-461beda2@172.30.0.168 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6313ms with no response
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK7ceddab7A46333B4
From: "4145" <sip:4145@customer.com>;tag=B6C579AB-89AD5B98
To: <sip:4119@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 4ed85f33-99e3da80-72a000c9@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="684fd3f8", uri="sip:4119@customer.com:5060", response="26ffdee5baf1a3221bea13cc113dd357", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (16 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:47:57] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK7ceddab7A46333B4;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=B6C579AB-89AD5B98
To: <sip:4119@customer.com>;tag=as7f9416bb
Call-ID: 4ed85f33-99e3da80-72a000c9@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5318f840"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog '4ed85f33-99e3da80-72a000c9@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] Retransmitting #6 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1202cbc7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4e261fea
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5de5c66b52b6d3c7210b2200441fbaa4@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:57] Really destroying SIP dialog '5de5c66b52b6d3c7210b2200441fbaa4@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:57] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK48329534
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as07c04e0f
To: "Office 106" <sip:4120@customer.com>;tag=783B04F3-1EAB67FC
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a016875-fdaf4e7e-42736207@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57] Retransmitting #6 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK65fbc5b4
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as46091936
To: "4145" <sip:4145@customer.com>;tag=CF3ED36-7B7B5347
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: 2645a66c-fc9dca45-6725fa2a@172.30.0.167
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK7dd1876d60DE438C
From: "4106" <sip:4106@customer.com>;tag=7AFFE505-8E3E0084
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: b86e3dc7-5211810e-25a67e11@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (15 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4106@172.30.0.171:5060>
[2011-11-03 06:47:57] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bK7dd1876d60DE438C;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=7AFFE505-8E3E0084
To: <sip:4108@customer.com>;tag=as14bed433
Call-ID: b86e3dc7-5211810e-25a67e11@172.30.0.171
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ba259f4"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog 'b86e3dc7-5211810e-25a67e11@172.30.0.171' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 2645a66c-fc9dca45-6725fa2a@172.30.0.167 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6421ms with no response
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.190:5060 --->
SUBSCRIBE sip:4130@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bKd73e27d5BE2B69E4
From: "Meeting Room" <sip:4130@customer.com>;tag=874C1147-960F01F6
To: <sip:4130@customer.com:5060>
CSeq: 1 SUBSCRIBE
Call-ID: 18535f23-983c4e92-8c852f31@172.30.0.190
Contact: <sip:4130@172.30.0.190:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (15 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.190:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4130@172.30.0.190:5060>
[2011-11-03 06:47:57] Found peer '4130' for '4130' from 172.30.0.190:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.190:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bKd73e27d5BE2B69E4;received=172.30.0.190
From: "Meeting Room" <sip:4130@customer.com>;tag=874C1147-960F01F6
To: <sip:4130@customer.com:5060>;tag=as00e919d8
Call-ID: 18535f23-983c4e92-8c852f31@172.30.0.190
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6508bf10"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog '18535f23-983c4e92-8c852f31@172.30.0.190' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] Really destroying SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.164' Method: REGISTER
[2011-11-03 06:47:57] Really destroying SIP dialog '8b77eefa-eedd49d-2bf144b4@172.30.0.168' Method: SUBSCRIBE
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK18f142497BDF6AF6
From: "4108" <sip:4108@customer.com>;tag=33C5AE9F-98B2FF24
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d50cf517-379c317c-661f2035@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (15 headers 0 lines) ---
[2011-11-03 06:47:57] Creating new subscription
[2011-11-03 06:47:57] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:47:57] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:47:57] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK18f142497BDF6AF6;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=33C5AE9F-98B2FF24
To: <sip:4122@customer.com>;tag=as4a2f0c1b
Call-ID: d50cf517-379c317c-661f2035@172.30.0.196
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="055a98e0"
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog 'd50cf517-379c317c-661f2035@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:57] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]
<--- SIP read from UDP:172.30.0.192:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK67edda0BD42FF2B
From: "4148" <sip:4148@customer.com>;tag=971A06C5-F38B6420
To: <sip:4148@customer.com>
CSeq: 2 REGISTER
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4148", realm="asterisk", nonce="4a0346ab", uri="sip:customer.com:5060", response="640a1773f3dfced62b9b824c0e76a351", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:57] --- (13 headers 0 lines) ---
[2011-11-03 06:47:57] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:47:57]     -- Registered SIP '4148' at 172.30.0.192:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK276fd2ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1fd4c08d
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f8dc7c62b34d8c501ed01194a191eff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:57]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bK67edda0BD42FF2B;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=971A06C5-F38B6420
To: <sip:4148@customer.com>;tag=as51f450d1
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.192
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.192:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:47:57 GMT
Content-Length: 0


<------------>
[2011-11-03 06:47:57] Scheduling destruction of SIP dialog '3b0be151-a656316c-99d39b7@172.30.0.192' in 32000 ms (Method: REGISTER)
[2011-11-03 06:47:57] Really destroying SIP dialog 'a884d693-fa67221a-d2c9b479@172.30.0.188' Method: SUBSCRIBE
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4119 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4142 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4116 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4128 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4125 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4137 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:47:57] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19aa9e2b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4136
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4148 (queued)
[2011-11-03 06:47:57] set_destination: Parsing <sip:4145@172.30.0.167:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.167:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3add5ebd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4145
[2011-11-03 06:47:57] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533fe5f6
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4113
[2011-11-03 06:47:57] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23ed3909
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4118
[2011-11-03 06:47:57] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK088093cd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4120
[2011-11-03 06:47:57] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54133146
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4119
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:47:57] set_destination: Parsing <sip:4106@172.30.0.171:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.171:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fd046b7
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4106
[2011-11-03 06:47:57] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:47:57] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:47:57] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2205b815
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4116
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4115 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4141 (queued)
[2011-11-03 06:47:57]   == Extension Changed 4148[default-local] new state Idle for Notify User 4147 (queued)
[2011-11-03 06:47:57] Retransmitting #3 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b51d386
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as094a02b4
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4946cd7224ccf4901ee58df76f84ffee@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:57] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07707bac
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as3a063028
To: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK59cfe5d121BE6DEA
From: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="2e102873", uri="sip:4147@customer.com:5060", response="1f373ac6153f4c3e29e30e70192a9a52", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:58] --- (16 headers 0 lines) ---
[2011-11-03 06:47:58] Creating new subscription
[2011-11-03 06:47:58] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:47:58] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:47:58] Looking for 4147 in default-local (domain customer.com:5060)
[2011-11-03 06:47:58] Scheduling destruction of SIP dialog '3a0ed75b-16468bc-154fc1f1@172.30.0.195' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:58]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK59cfe5d121BE6DEA;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
To: <sip:4147@customer.com>;tag=as746a85fa
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4147@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:58] set_destination: Parsing <sip:4113@172.30.0.195:5060> for address/port to send to
[2011-11-03 06:47:58] set_destination: set destination to 172.30.0.195:5060
[2011-11-03 06:47:58] Reliably Transmitting (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2055bcc0
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as746a85fa
To: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #3 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK67d60ed2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as77df2ea8
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62ad788e2212ca57369e7f2b53994167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:58] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1206b3ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4ee4835b
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 526bd48525ebf28d60c7c699612083fb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3add5ebd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533fe5f6
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d88f9ec
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK71fd35c3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Really destroying SIP dialog '2ff02f7b-7c7d8f8c-62981375@172.30.0.176' Method: SUBSCRIBE
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK088093cd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23ed3909
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKa76923a7AEA30B36
From: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4134", realm="asterisk", nonce="24078e2c", uri="sip:4147@customer.com:5060", response="f61eeb5accf8fe617bb76a36d3826b82", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:58] --- (16 headers 0 lines) ---
[2011-11-03 06:47:58] Creating new subscription
[2011-11-03 06:47:58] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:47:58] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:47:58] Looking for 4147 in default-local (domain customer.com:5060)
[2011-11-03 06:47:58] Scheduling destruction of SIP dialog 'c5f32c09-76ce4e08-4f93254f@172.30.0.179' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:58]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKa76923a7AEA30B36;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as397e6250
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4147@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:58] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:47:58] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:47:58] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07057c98
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as397e6250
To: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #5 (no NAT) to 172.30.0.184:5060:
NOTIFY sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcbd0a0
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as1bf9566b
To: "4123" <sip:4123@customer.com>;tag=5B819BBC-1AF518F5
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 6b074147-5962c75c-3c206d15@172.30.0.184
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4123@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Really destroying SIP dialog '973fd306-61221967-8836730c@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:47:58] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06674834
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0b3df67b
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7555d65d29f62798040af4023af79a54@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:58] Really destroying SIP dialog '7555d65d29f62798040af4023af79a54@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2055bcc0
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as746a85fa
To: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #2 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3add5ebd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58]
<--- SIP read from UDP:172.30.0.166:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK831c241aC84A2845
From: "4137" <sip:4137@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e56f2582-f4496c2d-8cab4c08@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4137", realm="asterisk", nonce="3061e589", uri="sip:4122@customer.com:5060", response="503070e026ad60de086470e02395397b", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:58] --- (16 headers 0 lines) ---
[2011-11-03 06:47:58] Creating new subscription
[2011-11-03 06:47:58] Sending to 172.30.0.166:5060 (no NAT)
[2011-11-03 06:47:58] list_route: hop: <sip:4137@172.30.0.166:5060>
[2011-11-03 06:47:58] Found peer '4137' for '4137' from 172.30.0.166:5060
[2011-11-03 06:47:58]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK831c241aC84A2845;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=AC052B50-19938FDB
To: <sip:4122@customer.com>;tag=as0444a95e
Call-ID: e56f2582-f4496c2d-8cab4c08@172.30.0.166
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="22126cc5"
Content-Length: 0


<------------>
[2011-11-03 06:47:58] Scheduling destruction of SIP dialog 'e56f2582-f4496c2d-8cab4c08@172.30.0.166' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:58] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533fe5f6
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54133146
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Really destroying SIP dialog '7f8bf698-7add0bfd-1a758a86@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:47:58] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78841311
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Really destroying SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:47:58] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK088093cd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19aa9e2b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #2 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2055bcc0
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as746a85fa
To: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23ed3909
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3a016875-fdaf4e7e-42736207@172.30.0.180 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 9913ms with no response
[2011-11-03 06:47:58] Retransmitting #5 (no NAT) to 172.30.0.196:5060:
NOTIFY sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4783d2c2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as53f2e80f
To: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4108@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #5 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d4d4415
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as32698f53
To: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKd712de335E4C9262
From: "Office 107" <sip:4141@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 197fee59-47d4d9d8-89034e9f@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:58] --- (15 headers 0 lines) ---
[2011-11-03 06:47:58] Creating new subscription
[2011-11-03 06:47:58] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:47:58] list_route: hop: <sip:4141@172.30.0.181:5060>
[2011-11-03 06:47:58] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:47:58]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKd712de335E4C9262;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as5653b4b3
Call-ID: 197fee59-47d4d9d8-89034e9f@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="047dc81e"
Content-Length: 0


<------------>
[2011-11-03 06:47:58] Scheduling destruction of SIP dialog '197fee59-47d4d9d8-89034e9f@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:58] Retransmitting #3 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3add5ebd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533fe5f6
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Reliably Transmitting (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3699a977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0bf6db91
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ed68f795b8df102565ddc9e1269d3af@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:58] Really destroying SIP dialog 'db6f3505-b4cfe424-a9e0f78b@172.30.0.199' Method: SUBSCRIBE
[2011-11-03 06:47:58]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK47948286A3EB9DF9
From: "4125" <sip:4125@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 8b77eefa-eedd49d-2bf144b4@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4125", realm="asterisk", nonce="231ddffd", uri="sip:4147@customer.com:5060", response="5bb13813e2b3f49fa444cde7ce26f566", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:58] --- (16 headers 0 lines) ---
[2011-11-03 06:47:58] Creating new subscription
[2011-11-03 06:47:58] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:47:58] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:47:58] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:47:58]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK47948286A3EB9DF9;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>;tag=as4b753617
Call-ID: 8b77eefa-eedd49d-2bf144b4@172.30.0.168
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ae6e5d0"
Content-Length: 0


<------------>
[2011-11-03 06:47:58] Scheduling destruction of SIP dialog '8b77eefa-eedd49d-2bf144b4@172.30.0.168' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:58] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKd50400c2120071B9
From: "4116" <sip:4116@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="6dba1b29", uri="sip:4115@customer.com:5060", response="b3933f632816f57f0e6078b925cee770", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:58] --- (16 headers 0 lines) ---
[2011-11-03 06:47:58] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:58] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:47:58] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4116" <sip:4116@customer.com>;tag=CC75A1B6-F8970A0D'
[2011-11-03 06:47:58]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKd50400c2120071B9;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as4f0abc0f
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fbcbf17", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:47:58] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:58] Retransmitting #3 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2055bcc0
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as746a85fa
To: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #4 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b51d386
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as094a02b4
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4946cd7224ccf4901ee58df76f84ffee@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:58] Really destroying SIP dialog '4946cd7224ccf4901ee58df76f84ffee@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:58] Really destroying SIP dialog 'b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.183' Method: SUBSCRIBE
[2011-11-03 06:47:58] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54133146
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #6 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5628303c
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a415b7
To: "4125" <sip:4125@customer.com>;tag=8EF217BE-B0414651
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:58] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK276fd2ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1fd4c08d
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f8dc7c62b34d8c501ed01194a191eff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:58] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a02e6f59-4d240880-e0e88a0b@172.30.0.178 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6458ms with no response
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKdf2ca764E8EAD52B
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (15 headers 0 lines) ---
[2011-11-03 06:47:59] Creating new subscription
[2011-11-03 06:47:59] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:59] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:59] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:59]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKdf2ca764E8EAD52B;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>;tag=as1e18c6e3
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0dcdf702"
Content-Length: 0


<------------>
[2011-11-03 06:47:59] Scheduling destruction of SIP dialog '31caffa2-d80e65c9-ba5c76a0@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:59] Retransmitting #4 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #4 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK67d60ed2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as77df2ea8
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 62ad788e2212ca57369e7f2b53994167@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:59] Really destroying SIP dialog '62ad788e2212ca57369e7f2b53994167@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:59] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1206b3ed
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4ee4835b
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 526bd48525ebf28d60c7c699612083fb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:59] Really destroying SIP dialog '526bd48525ebf28d60c7c699612083fb@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:47:59] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25d27825
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 78031bfd-6cc68e94-cf2353cf@172.30.0.168 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6472ms with no response
[2011-11-03 06:47:59] Retransmitting #5 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #6 (no NAT) to 172.30.0.165:5060:
NOTIFY sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0ad70ab7
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as159032dc
To: "Office 113" <sip:4171@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: d2df29bd-7b13dd2-3f183dd3@172.30.0.165
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4171@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK088093cd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK62debc256558189A
From: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="732d40f2", uri="sip:4108@customer.com:5060", response="c3ed5f5b790d40977aa5d977e483c3c0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (16 headers 0 lines) ---
[2011-11-03 06:47:59] Creating new subscription
[2011-11-03 06:47:59] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:47:59] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:47:59] Looking for 4108 in default-local (domain customer.com:5060)
[2011-11-03 06:47:59] Scheduling destruction of SIP dialog '569c1668-7cf1ab91-1b38956@172.30.0.177' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:59]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK62debc256558189A;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
To: <sip:4108@customer.com>;tag=as0a17f176
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4108@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:59] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:47:59] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:47:59] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dc953f5
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as0a17f176
To: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Really destroying SIP dialog 'db6bcec2-2e8acd45-563892bc@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:47:59] Really destroying SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.179' Method: REGISTER
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission d2df29bd-7b13dd2-3f183dd3@172.30.0.165 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6425ms with no response
[2011-11-03 06:47:59] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23ed3909
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK2b8a60b569C1668
From: "4138" <sip:4138@customer.com>;tag=902627F9-2BA4261E
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7cf1ab91-1b38956-1e32847@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (15 headers 0 lines) ---
[2011-11-03 06:47:59] Creating new subscription
[2011-11-03 06:47:59] Sending to 172.30.0.163:5060 (no NAT)
[2011-11-03 06:47:59] list_route: hop: <sip:4138@172.30.0.163:5060>
[2011-11-03 06:47:59] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:47:59]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK2b8a60b569C1668;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=902627F9-2BA4261E
To: <sip:4108@customer.com>;tag=as7c8bbd9e
Call-ID: 7cf1ab91-1b38956-1e32847@172.30.0.163
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ec7e226"
Content-Length: 0


<------------>
[2011-11-03 06:47:59] Scheduling destruction of SIP dialog '7cf1ab91-1b38956-1e32847@172.30.0.163' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:59] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9261d0
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as252ea80a
To: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b0ba94bb-b41a10b8-a3494831@172.30.0.168 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2011-11-03 06:47:59] Retransmitting #12 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #6 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c484883
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as127a9232
To: "4138" <sip:4138@customer.com>;tag=C47CD70A-391C653F
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6351d728-8b1f2125-2caa38e@172.30.0.163
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #6 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0b31f9f3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1df1c6ea
To: "4135" <sip:4135@customer.com>;tag=1401F2A3-2920F280
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: b0ba94bb-b41a10b8-a3494831@172.30.0.177
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: terminated;reason=timeout
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #6 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4886abd9
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as66a5e48b
To: "4145" <sip:4145@customer.com>;tag=7B9E7168-8CB5B591
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 4f79f2-a0609d53-c7e55e20@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #6 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK739c3275
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fa3ce12
To: "Office 131" <sip:4113@customer.com>;tag=22FC91F9-618AB732
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 6986012d-1e6eb376-e42a19b3@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK9708841f46781D2C
From: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (15 headers 0 lines) ---
[2011-11-03 06:47:59] Ignoring this SUBSCRIBE request
[2011-11-03 06:47:59] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:47:59]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK9708841f46781D2C;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as74380269
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01949029"
Content-Length: 0


<------------>
[2011-11-03 06:47:59] Scheduling destruction of SIP dialog '65ddf445-1153d83a-5a70009b@172.30.0.177' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 6351d728-8b1f2125-2caa38e@172.30.0.163 for seqno 105 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6538ms with no response
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b0ba94bb-b41a10b8-a3494831@172.30.0.177 for seqno 111 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6531ms with no response
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 4f79f2-a0609d53-c7e55e20@172.30.0.167 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6523ms with no response
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 6986012d-1e6eb376-e42a19b3@172.30.0.195 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6519ms with no response
[2011-11-03 06:47:59] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK324b0da5
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as773db5d8
To: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #6 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa188c07130CA207A
From: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
To: <sip:4140@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="02ae16f8", uri="sip:4140@customer.com:5060", response="7e91f06b2505875c1f4c0013256e5a6b", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (16 headers 0 lines) ---
[2011-11-03 06:47:59] Creating new subscription
[2011-11-03 06:47:59] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:47:59] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:47:59] Looking for 4140 in default-local (domain customer.com:5060)
[2011-11-03 06:47:59] Scheduling destruction of SIP dialog 'a088c697-bbc01fa0-e89a2629@172.30.0.180' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:59]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKa188c07130CA207A;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
To: <sip:4140@customer.com>;tag=as00e218e3
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4140@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:47:59] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:47:59] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:47:59] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35210e4e
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as00e218e3
To: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19aa9e2b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #4 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3add5ebd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533fe5f6
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a02e6f59-4d240880-e0e88a0b@172.30.0.178 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6473ms with no response
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19f6d50b
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
CSeq: 102 NOTIFY
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (11 headers 0 lines) ---
[2011-11-03 06:47:59] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:47:59] set_destination: Parsing <sip:4141@172.30.0.181:5060> for address/port to send to
[2011-11-03 06:47:59] set_destination: set destination to 172.30.0.181:5060
[2011-11-03 06:47:59] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f95cdd5
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59]   == Extension Changed 4148[default-local] new state Idle for Notify User 4141
[2011-11-03 06:47:59] Retransmitting #4 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7fe241bb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Really destroying SIP dialog 'ebf68d48-834d5031-6928867e@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:47:59] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dc953f5
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as0a17f176
To: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #4 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2055bcc0
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as746a85fa
To: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Really destroying SIP dialog '1da9150f-2b3ef580-8d99cf05@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:47:59] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35210e4e
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as00e218e3
To: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #10 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Retransmitting #7 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4106@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK8239d127EA2F445E
From: "4118" <sip:4118@customer.com>;tag=719AEFC5-F897A95C
To: <sip:4106@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: aea0cf45-c854a0dc-bfaa1823@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (15 headers 0 lines) ---
[2011-11-03 06:47:59] Creating new subscription
[2011-11-03 06:47:59] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:47:59] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:47:59] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:47:59]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK8239d127EA2F445E;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=719AEFC5-F897A95C
To: <sip:4106@customer.com>;tag=as77e202ba
Call-ID: aea0cf45-c854a0dc-bfaa1823@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5933db51"
Content-Length: 0


<------------>
[2011-11-03 06:47:59] Scheduling destruction of SIP dialog 'aea0cf45-c854a0dc-bfaa1823@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:59] Retransmitting #1 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3699a977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0bf6db91
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ed68f795b8df102565ddc9e1269d3af@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:47:59] Retransmitting #1 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07057c98
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as397e6250
To: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:47:59] Really destroying SIP dialog 'b86e3dc7-5211810e-25a67e11@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:47:59]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bKdf9491cbF3731D5C
From: "Office 108" <sip:4128@customer.com>;tag=CCEF9FC3-1FFD0894
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:47:59] --- (15 headers 0 lines) ---
[2011-11-03 06:47:59] Creating new subscription
[2011-11-03 06:47:59] Sending to 172.30.0.182:5060 (no NAT)
[2011-11-03 06:47:59] list_route: hop: <sip:4128@172.30.0.182:5060>
[2011-11-03 06:47:59] Found peer '4128' for '4128' from 172.30.0.182:5060
[2011-11-03 06:47:59]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bKdf9491cbF3731D5C;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=CCEF9FC3-1FFD0894
To: <sip:4147@customer.com>;tag=as2a0f8c77
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27dbd119"
Content-Length: 0


<------------>
[2011-11-03 06:47:59] Scheduling destruction of SIP dialog 'b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:47:59] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK276fd2ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1fd4c08d
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f8dc7c62b34d8c501ed01194a191eff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.189:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK61bda54dD0C60664
From: "Boardroom" <sip:4109@customer.com>;tag=2681885B-184EF31A
To: <sip:4109@customer.com:5060>
CSeq: 1 SUBSCRIBE
Call-ID: a756ff0f-26f3e83e-55482e21@172.30.0.189
Contact: <sip:4109@172.30.0.189:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (15 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.189:5060 (no NAT)
[2011-11-03 06:48:00] list_route: hop: <sip:4109@172.30.0.189:5060>
[2011-11-03 06:48:00] Found peer '4109' for '4109' from 172.30.0.189:5060
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.189:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK61bda54dD0C60664;received=172.30.0.189
From: "Boardroom" <sip:4109@customer.com>;tag=2681885B-184EF31A
To: <sip:4109@customer.com:5060>;tag=as05b18d87
Call-ID: a756ff0f-26f3e83e-55482e21@172.30.0.189
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a78ef50"
Content-Length: 0


<------------>
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog 'a756ff0f-26f3e83e-55482e21@172.30.0.189' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00] Retransmitting #5 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35210e4e
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as00e218e3
To: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54133146
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Reliably Transmitting (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36762d5b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37e5b7bd
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49462d9a03fa9094643793f873b2bdef@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:00] Retransmitting #6 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06f91a0a
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as21c64c30
To: "4116" <sip:4116@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 76ce4e08-4f93254f-34c24d9e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK125f41e0
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
CSeq: 102 NOTIFY
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (11 headers 0 lines) ---
[2011-11-03 06:48:00] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 76ce4e08-4f93254f-34c24d9e@172.30.0.199 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6413ms with no response
[2011-11-03 06:48:00] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK088093cd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07707bac
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as3a063028
To: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2205b815
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.178:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2d3de8ae
From: <sip:4127@customer.com>;tag=as1ce2bd26
To: "4173" <sip:4173@customer.com>;tag=1A5028C3-D00A2A42
CSeq: 103 NOTIFY
Call-ID: a02e6f59-4d240880-e0e88a0b@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (11 headers 0 lines) ---
[2011-11-03 06:48:00] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47a91343
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Really destroying SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.199' Method: REGISTER
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.166:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK8610df405EB7CACB
From: "4137" <sip:4137@customer.com>;tag=64514EDD-574841B8
To: <sip:4148@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 8c28ad86-c4c5a271-4bc8f48c@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (15 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.166:5060 (no NAT)
[2011-11-03 06:48:00] list_route: hop: <sip:4137@172.30.0.166:5060>
[2011-11-03 06:48:00] Found peer '4137' for '4137' from 172.30.0.166:5060
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK8610df405EB7CACB;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=64514EDD-574841B8
To: <sip:4148@customer.com>;tag=as717d8f4d
Call-ID: 8c28ad86-c4c5a271-4bc8f48c@172.30.0.166
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6caaa015"
Content-Length: 0


<------------>
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog '8c28ad86-c4c5a271-4bc8f48c@172.30.0.166' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00] Really destroying SIP dialog 'e8fa9348-b82111f1-f5635466@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:48:00] Reliably Transmitting (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d5004a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4aeb60e4
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7f78d35e27cb790e2911e7437e5d9ccb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:00] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23ed3909
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #7 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK4f93254f34C24D9E
From: "Office 107" <sip:4141@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: bdf5d4b5-60ff2a54-bae9903b@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (15 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:48:00] list_route: hop: <sip:4141@172.30.0.181:5060>
[2011-11-03 06:48:00] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK4f93254f34C24D9E;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as546cd388
Call-ID: bdf5d4b5-60ff2a54-bae9903b@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15bcde7c"
Content-Length: 0


<------------>
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog 'bdf5d4b5-60ff2a54-bae9903b@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00] Really destroying SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.171' Method: REGISTER
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK1153d83a5A70009B
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (15 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.163:5060 (no NAT)
[2011-11-03 06:48:00] list_route: hop: <sip:4138@172.30.0.163:5060>
[2011-11-03 06:48:00] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK1153d83a5A70009B;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as3c290715
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a6bd63c"
Content-Length: 0


<------------>
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog 'ddf9dab8-ab9de2a1-75ab5826@172.30.0.163' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00] Really destroying SIP dialog '4f4a86e4-c1c0933d-d5d4f48a@172.30.0.184' Method: SUBSCRIBE
[2011-11-03 06:48:00] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35210e4e
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as00e218e3
To: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dc953f5
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as0a17f176
To: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKe90f478d3644186C
From: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="7ceefd6a", uri="sip:4147@customer.com:5060", response="570daa9e220ea1a92895acf2ae84b13f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (16 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:48:00] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:48:00] Looking for 4147 in default-local (domain customer.com:5060)
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog 'c5f32c09-76ce4e08-4f93254f@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKe90f478d3644186C;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
To: <sip:4147@customer.com>;tag=as24d6fe6e
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4147@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:00] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:48:00] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:48:00] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b272693
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as24d6fe6e
To: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Really destroying SIP dialog 'ab321eb1-229b970-7cbd5377@172.30.0.181' Method: REGISTER
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK6288baf6B4C5C4A9
From: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4125", realm="asterisk", nonce="3ed8ef53", uri="sip:4148@customer.com:5060", response="4eff9547404eee6ba324fc906ba1b05f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (16 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:48:00] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:48:00] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog 'e47da7d6-f2289e09-ce590c70@172.30.0.168' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK6288baf6B4C5C4A9;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
To: <sip:4148@customer.com>;tag=as7b42bfd5
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:00] set_destination: Parsing <sip:4125@172.30.0.168:5060> for address/port to send to
[2011-11-03 06:48:00] set_destination: set destination to 172.30.0.168:5060
[2011-11-03 06:48:00] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ff6416e
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7b42bfd5
To: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fd046b7
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #2 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3699a977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0bf6db91
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ed68f795b8df102565ddc9e1269d3af@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:00] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a17c643
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca4208d
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 405621e02c094e0636cd9e370b156f73@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK6f0cee3c8442B2E3
From: "Office 107" <sip:4141@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c1452592-c5f32c09-76ce4e08@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (15 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:48:00] list_route: hop: <sip:4141@172.30.0.181:5060>
[2011-11-03 06:48:00] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK6f0cee3c8442B2E3;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as21157159
Call-ID: c1452592-c5f32c09-76ce4e08@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15cad2e1"
Content-Length: 0


<------------>
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog 'c1452592-c5f32c09-76ce4e08@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKddf9dab8AB9DE2A1
From: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (15 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:48:00] list_route: hop: <sip:4135@172.30.0.177:5060>
[2011-11-03 06:48:00] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKddf9dab8AB9DE2A1;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>;tag=as440eeb94
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d6f9db2"
Content-Length: 0


<------------>
[2011-11-03 06:48:00] Scheduling destruction of SIP dialog '75ab5826-6d5f5fd7-bf46e004@172.30.0.177' in 32000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:00] Really destroying SIP dialog '43c45281-77964ac0-3ae67647@172.30.0.199' Method: SUBSCRIBE
[2011-11-03 06:48:00] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ff6416e
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7b42bfd5
To: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:00] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK276fd2ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1fd4c08d
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f8dc7c62b34d8c501ed01194a191eff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:00]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4136@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK5373d403F7EFF10
From: "4145" <sip:4145@customer.com>;tag=4B97035A-1372D29B
To: <sip:4136@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: e8fa9348-b82111f1-f5635466@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4145", realm="asterisk", nonce="2170ec07", uri="sip:4136@customer.com:5060", response="06408b35654f64e4e3d2306ea9b2278a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:00] --- (16 headers 0 lines) ---
[2011-11-03 06:48:00] Creating new subscription
[2011-11-03 06:48:00] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:48:00] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:48:00] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:48:00]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK5373d403F7EFF10;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=4B97035A-1372D29B
To: <sip:4136@customer.com>;tag=as3a10f99f
Call-ID: e8fa9348-b82111f1-f5635466@172.30.0.167
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f1ab1b6"
Content-Length: 0


<------------>
[2011-11-03 06:48:01] Scheduling destruction of SIP dialog 'e8fa9348-b82111f1-f5635466@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:01] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b272693
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as24d6fe6e
To: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #5 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3add5ebd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533fe5f6
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5d88f9ec
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
CSeq: 103 NOTIFY
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:48:01] --- (11 headers 0 lines) ---
[2011-11-03 06:48:01] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:48:01] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ff6416e
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7b42bfd5
To: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKc0af2da2C316772B
From: "Office 106" <sip:4120@customer.com>;tag=42BDEE3D-C7CD8046
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 35317d09-75c0c112-fd33329b@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="5e895934", uri="sip:4108@customer.com:5060", response="7d5f027361eb64c716f98d205e0e736f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:01] --- (16 headers 0 lines) ---
[2011-11-03 06:48:01] Creating new subscription
[2011-11-03 06:48:01] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:48:01] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:48:01] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:48:01]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKc0af2da2C316772B;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=42BDEE3D-C7CD8046
To: <sip:4108@customer.com>;tag=as317802ec
Call-ID: 35317d09-75c0c112-fd33329b@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d87e8fd"
Content-Length: 0


<------------>
[2011-11-03 06:48:01] Scheduling destruction of SIP dialog '35317d09-75c0c112-fd33329b@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:01] Reliably Transmitting (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK021bbf7b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5d198f08
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0cdf6180063048001b065f53446adef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01]
<--- SIP read from UDP:172.30.0.166:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK9f3dfab6424F6AA1
From: "4137" <sip:4137@customer.com>;tag=971A06C5-F38B6420
To: <sip:4137@customer.com>
CSeq: 2 REGISTER
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
ser-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4137", realm="asterisk", nonce="331db230", uri="sip:customer.com:5060", response="88302f2ba69e5efd3c23cb760280d78c", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:01] --- (13 headers 0 lines) ---
[2011-11-03 06:48:01] Sending to 172.30.0.166:5060 (no NAT)
[2011-11-03 06:48:01] Reliably Transmitting (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25a94984
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c9d92ee
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5a9aa32d3f01afce31fda86b1907ba9e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bK9f3dfab6424F6AA1;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=971A06C5-F38B6420
To: <sip:4137@customer.com>;tag=as6517afda
Call-ID: 3b0be151-a656316c-99d39b7@172.30.0.166
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4137@172.30.0.166:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:48:01 GMT
Content-Length: 0


<------------>
[2011-11-03 06:48:01] Scheduling destruction of SIP dialog '3b0be151-a656316c-99d39b7@172.30.0.166' in 32000 ms (Method: REGISTER)
[2011-11-03 06:48:01] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK49e3cb04
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6e50f4
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ed9ad4b4383baeb5c442b0b014c159d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ec55a50
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a638217
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4a7586b4029ac9ec5d6136e87fd53b39@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01] Retransmitting #1 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36762d5b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37e5b7bd
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49462d9a03fa9094643793f873b2bdef@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01] Retransmitting #5 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2055bcc0
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as746a85fa
To: "Office 131" <sip:4113@customer.com>;tag=13CA678D-8A943156
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: 3a0ed75b-16468bc-154fc1f1@172.30.0.195
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #14 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7069a5d3
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as6c14ff6a
To: "4148" <sip:4148@customer.com>;tag=C1793082-6D0E71E9
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 6b074147-5962c75c-3c206d15@172.30.0.184 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6703ms with no response
[2011-11-03 06:48:01] Retransmitting #7 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #5 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19aa9e2b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ff6416e
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7b42bfd5
To: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #1 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d5004a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4aeb60e4
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7f78d35e27cb790e2911e7437e5d9ccb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01] Really destroying SIP dialog '32de0b9d-c79fec9a-7fe39c4b@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:48:01] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b272693
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as24d6fe6e
To: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 834d5031-6928867e-29412cbf@172.30.0.196 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6384ms with no response
[2011-11-03 06:48:01]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK3644186cC1A70F93
From: "4124" <sip:4124@customer.com>;tag=AC291EE5-39D6A704
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d50400c2-120071b9-35e78a38@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:01] --- (15 headers 0 lines) ---
[2011-11-03 06:48:01] Creating new subscription
[2011-11-03 06:48:01] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:48:01] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:48:01] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:48:01]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK3644186cC1A70F93;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=AC291EE5-39D6A704
To: <sip:4140@customer.com>;tag=as31837346
Call-ID: d50400c2-120071b9-35e78a38@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7414de83"
Content-Length: 0


<------------>
[2011-11-03 06:48:01] Scheduling destruction of SIP dialog 'd50400c2-120071b9-35e78a38@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:01] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #3 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3699a977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0bf6db91
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ed68f795b8df102565ddc9e1269d3af@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d4d4415
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as32698f53
To: "4147" <sip:4147@customer.com>;tag=88DB6959-26E871BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 198759bc-5af6fe25-2cc894a6@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a17c643
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca4208d
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 405621e02c094e0636cd9e370b156f73@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35210e4e
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as00e218e3
To: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:01]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKf1dc0a9cBA4BF443
From: "4116" <sip:4116@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 43c45281-77964ac0-3ae67647@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="5b8e9866", uri="sip:4127@customer.com:5060", response="406365e3c4a2cbed1d93350b46bd29cf", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:01] --- (16 headers 0 lines) ---
[2011-11-03 06:48:01] Creating new subscription
[2011-11-03 06:48:01] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:48:01] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-11-03 06:48:01] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:48:01]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKf1dc0a9cBA4BF443;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as68b7ccfc
Call-ID: 43c45281-77964ac0-3ae67647@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f9c7fb8"
Content-Length: 0


<------------>
[2011-11-03 06:48:01] Scheduling destruction of SIP dialog '43c45281-77964ac0-3ae67647@172.30.0.199' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:01] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 198759bc-5af6fe25-2cc894a6@172.30.0.183 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6459ms with no response
[2011-11-03 06:48:01] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
OPTIONS sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK276fd2ad
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as1fd4c08d
To: <sip:4148@172.30.0.192:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6f8dc7c62b34d8c501ed01194a191eff@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:01] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4148' is now UNREACHABLE!  Last qualify: 0
[2011-11-03 06:48:01] Really destroying SIP dialog '6f8dc7c62b34d8c501ed01194a191eff@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4142 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4116 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4128 (queued)
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:48:01] Really destroying SIP dialog '13d339de-a62eda3b-c4eedd1c@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:48:01]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4125 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4137 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4148 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4113 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4119 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4106 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4116 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4115 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4141 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4147 (queued)
[2011-11-03 06:48:02]   == Extension Changed 4148[default-local] new state Unavailable for Notify User 4125 (queued)
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.199:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKbdf5d4b560FF2A54
From: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4116", realm="asterisk", nonce="41dff001", uri="sip:4122@customer.com:5060", response="04ef110c40b5e0a28b666f68fce113f0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (16 headers 0 lines) ---
[2011-11-03 06:48:02] Creating new subscription
[2011-11-03 06:48:02] Sending to 172.30.0.199:5060 (no NAT)
[2011-11-03 06:48:02] Found peer '4116' for '4116' from 172.30.0.199:5060
[2011-11-03 06:48:02] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog '8eca8c0c-d712de33-5e4c9262@172.30.0.199' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.199:5060;branch=z9hG4bKbdf5d4b560FF2A54;received=172.30.0.199
From: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as2eff3dab
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:02] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-11-03 06:48:02] set_destination: set destination to 172.30.0.199:5060
[2011-11-03 06:48:02] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a1eff0c
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2eff3dab
To: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK71fd35c3
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as4fe58b4c
To: "4118" <sip:4118@customer.com>;tag=F95DFBDD-140540F4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 2e56d757-73b8c18e-38727375@172.30.0.173
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4106@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK8239d127EA2F445E
From: "4118" <sip:4118@customer.com>;tag=719AEFC5-F897A95C
To: <sip:4106@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: aea0cf45-c854a0dc-bfaa1823@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (15 headers 0 lines) ---
[2011-11-03 06:48:02] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:02] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK8239d127EA2F445E;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=719AEFC5-F897A95C
To: <sip:4106@customer.com>;tag=as77e202ba
Call-ID: aea0cf45-c854a0dc-bfaa1823@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5933db51"
Content-Length: 0


<------------>
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog 'aea0cf45-c854a0dc-bfaa1823@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK021bbf7b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5d198f08
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0cdf6180063048001b065f53446adef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02] Retransmitting #2 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36762d5b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37e5b7bd
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49462d9a03fa9094643793f873b2bdef@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK36c90081ACD81686
From: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="01949029", uri="sip:4109@customer.com:5060", response="e943c86544fd29201f910b0e2a001fb8", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (16 headers 0 lines) ---
[2011-11-03 06:48:02] Creating new subscription
[2011-11-03 06:48:02] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:48:02] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:48:02] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog '65ddf445-1153d83a-5a70009b@172.30.0.177' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK36c90081ACD81686;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as74380269
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:02] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:48:02] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:48:02] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK55a1ee55
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as74380269
To: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a1eff0c
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2eff3dab
To: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #6 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ff6416e
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7b42bfd5
To: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc716d2fa72870C31
From: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (15 headers 0 lines) ---
[2011-11-03 06:48:02] Creating new subscription
[2011-11-03 06:48:02] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:48:02] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:48:02] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc716d2fa72870C31;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as0135ac39
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6fb0812a"
Content-Length: 0


<------------>
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78841311
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK861346e67B9424C7
From: "4140" <sip:4140@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 32de0b9d-c79fec9a-7fe39c4b@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4140", realm="asterisk", nonce="61ba09ec", uri="sip:4108@customer.com:5060", response="612ac6839f6a0687e63e310f4c59cba0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (16 headers 0 lines) ---
[2011-11-03 06:48:02] Creating new subscription
[2011-11-03 06:48:02] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:48:02] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:48:02] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK861346e67B9424C7;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4108@customer.com>;tag=as3096defb
Call-ID: 32de0b9d-c79fec9a-7fe39c4b@172.30.0.175
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4141ebeb"
Content-Length: 0


<------------>
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog '32de0b9d-c79fec9a-7fe39c4b@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02] Really destroying SIP dialog 'c79fec9a-7fe39c4b-fb985080@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:48:02] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54133146
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a1eff0c
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2eff3dab
To: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.190:5060 --->
SUBSCRIBE sip:4130@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bKd73e27d5BE2B69E4
From: "Meeting Room" <sip:4130@customer.com>;tag=874C1147-960F01F6
To: <sip:4130@customer.com:5060>
CSeq: 1 SUBSCRIBE
Call-ID: 18535f23-983c4e92-8c852f31@172.30.0.190
Contact: <sip:4130@172.30.0.190:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: missed-call-summary
User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: message/sipfrag
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (15 headers 0 lines) ---
[2011-11-03 06:48:02] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:02] Found peer '4130' for '4130' from 172.30.0.190:5060
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.190:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.190:5060;branch=z9hG4bKd73e27d5BE2B69E4;received=172.30.0.190
From: "Meeting Room" <sip:4130@customer.com>;tag=874C1147-960F01F6
To: <sip:4130@customer.com:5060>;tag=as00e919d8
Call-ID: 18535f23-983c4e92-8c852f31@172.30.0.190
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6508bf10"
Content-Length: 0


<------------>
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog '18535f23-983c4e92-8c852f31@172.30.0.190' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02] Retransmitting #2 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d5004a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4aeb60e4
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7f78d35e27cb790e2911e7437e5d9ccb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02] Retransmitting #6 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.196:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4783d2c2
From: <sip:4109@customer.com>;tag=as53f2e80f
To: "4108" <sip:4108@customer.com>;tag=9E927EE5-F7526B02
CSeq: 102 NOTIFY
Call-ID: 834d5031-6928867e-29412cbf@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (11 headers 0 lines) ---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK1a520a2188246C90
From: "Office 105" <sip:4131@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4131", realm="asterisk", nonce="51037460", uri="sip:4122@customer.com:5060", response="01717368abb00e9b5a99b69269bf1c31", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (16 headers 0 lines) ---
[2011-11-03 06:48:02] Creating new subscription
[2011-11-03 06:48:02] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:48:02] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:48:02] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK1a520a2188246C90;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>;tag=as52503a50
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.170
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="55ad0ea6"
Content-Length: 0


<------------>
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dc953f5
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as0a17f176
To: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK088093cd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25a94984
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c9d92ee
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5a9aa32d3f01afce31fda86b1907ba9e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.192:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bKc0e60aec6CAFEB37
From: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
Contact: <sip:4148@172.30.0.192:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4148", realm="asterisk", nonce="695fb4bc", uri="sip:4115@customer.com:5060", response="6b4475e4accb78e40a36ca356555cf2a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (16 headers 0 lines) ---
[2011-11-03 06:48:02] Creating new subscription
[2011-11-03 06:48:02] Sending to 172.30.0.192:5060 (no NAT)
[2011-11-03 06:48:02] Found peer '4148' for '4148' from 172.30.0.192:5060
[2011-11-03 06:48:02] Looking for 4115 in default-local (domain customer.com:5060)
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog '312cc0b9-ab7a3354-7ac3461f@172.30.0.192' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.192:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.192:5060;branch=z9hG4bKc0e60aec6CAFEB37;received=172.30.0.192
From: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
To: <sip:4115@customer.com>;tag=as3f9e7318
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4115@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:02] set_destination: Parsing <sip:4148@172.30.0.192:5060> for address/port to send to
[2011-11-03 06:48:02] set_destination: set destination to 172.30.0.192:5060
[2011-11-03 06:48:02] Reliably Transmitting (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27283f8b
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as3f9e7318
To: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Reliably Transmitting (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4484670a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2cc38e79
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 66a6a25600f7a6fa10d867476c2585ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK49e3cb04
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6e50f4
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ed9ad4b4383baeb5c442b0b014c159d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56aeb16c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7492b7ad
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6781810952ce9bf369c79c09674784fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ec55a50
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a638217
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4a7586b4029ac9ec5d6136e87fd53b39@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02] Retransmitting #4 (no NAT) to 172.30.0.170:5060:
OPTIONS sip:4131@172.30.0.170:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3699a977
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0bf6db91
To: <sip:4131@172.30.0.170:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 1ed68f795b8df102565ddc9e1269d3af@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:47:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02] Really destroying SIP dialog '1ed68f795b8df102565ddc9e1269d3af@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:02] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b272693
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as24d6fe6e
To: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a17c643
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca4208d
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 405621e02c094e0636cd9e370b156f73@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:02]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK861346e67B9424C7
From: "4140" <sip:4140@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 32de0b9d-c79fec9a-7fe39c4b@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4140", realm="asterisk", nonce="61ba09ec", uri="sip:4108@customer.com:5060", response="612ac6839f6a0687e63e310f4c59cba0", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:02] --- (16 headers 0 lines) ---
[2011-11-03 06:48:02] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:02] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:48:02] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"4140" <sip:4140@customer.com>;tag=9E927EE5-F7526B02'
[2011-11-03 06:48:02]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK861346e67B9424C7;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=9E927EE5-F7526B02
To: <sip:4108@customer.com>;tag=as3096defb
Call-ID: 32de0b9d-c79fec9a-7fe39c4b@172.30.0.175
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4141ebeb", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:48:02] Scheduling destruction of SIP dialog '32de0b9d-c79fec9a-7fe39c4b@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK55a1ee55
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as74380269
To: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f95cdd5
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a1eff0c
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2eff3dab
To: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #2 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07057c98
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as397e6250
To: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Really destroying SIP dialog '661f2035-7aecdd92-22db8323@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:48:02] Retransmitting #1 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27283f8b
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as3f9e7318
To: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:02] Retransmitting #5 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKfa2b9b047452D01F
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (15 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4115@172.30.0.197:5060>
[2011-11-03 06:48:03] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKfa2b9b047452D01F;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>;tag=as756b08e3
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5dbddde7"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog 'f76a8dc6-903f12a9-9987ba38@172.30.0.197' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25d27825
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #6 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKd712de335E4C9262
From: "Office 107" <sip:4141@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 197fee59-47d4d9d8-89034e9f@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (15 headers 0 lines) ---
[2011-11-03 06:48:03] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:03] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bKd712de335E4C9262;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=5FC2D8D4-741BAABB
To: <sip:4122@customer.com>;tag=as5653b4b3
Call-ID: 197fee59-47d4d9d8-89034e9f@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="047dc81e"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '197fee59-47d4d9d8-89034e9f@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Really destroying SIP dialog '7f0aa159-11595212-dcaaf37f@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.169:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12b3a6fe
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as42b0f452
To: "4127" <sip:4127@172.30.0.169:5060>;tag=86A3BBFB-8FD0A098
CSeq: 102 OPTIONS
Call-ID: 6514502c37c16ff455cdada32716a07b@172.30.0.10:5060
Contact: <sip:4127@172.30.0.169:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (14 headers 0 lines) ---
[2011-11-03 06:48:03] Retransmitting #2 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27283f8b
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as3f9e7318
To: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #5 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23ed3909
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9261d0
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as252ea80a
To: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25d27825
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
CSeq: 103 NOTIFY
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (11 headers 0 lines) ---
[2011-11-03 06:48:03] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:48:03] Retransmitting #13 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #2 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK021bbf7b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5d198f08
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0cdf6180063048001b065f53446adef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Retransmitting #3 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36762d5b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37e5b7bd
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49462d9a03fa9094643793f873b2bdef@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK324b0da5
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as773db5d8
To: "Office 106" <sip:4120@customer.com>;tag=6715768E-13D7A217
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 136b5158-e366ebe1-5d9773ea@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2941e310C96F1199
From: "Office 106" <sip:4120@customer.com>;tag=8235C922-63F652AB
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 2c10135c-cddf33e5-efec41ee@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="0cab4146", uri="sip:4122@customer.com:5060", response="26ac9b23c170b3e5c1d521c1ab1c62a6", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (16 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:48:03] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK2941e310C96F1199;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=8235C922-63F652AB
To: <sip:4122@customer.com>;tag=as30300944
Call-ID: 2c10135c-cddf33e5-efec41ee@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4bc854c2"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '2c10135c-cddf33e5-efec41ee@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKdf2ca764E8EAD52B
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (15 headers 0 lines) ---
[2011-11-03 06:48:03] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:03] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKdf2ca764E8EAD52B;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>;tag=as1e18c6e3
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0dcdf702"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '31caffa2-d80e65c9-ba5c76a0@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 2e56d757-73b8c18e-38727375@172.30.0.173 for seqno 104 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10656ms with no response
[2011-11-03 06:48:03] Retransmitting #3 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d5004a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4aeb60e4
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7f78d35e27cb790e2911e7437e5d9ccb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Retransmitting #3 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27283f8b
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as3f9e7318
To: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK84fda8269853BA89
From: "4115" <sip:4115@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (15 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4115@172.30.0.197:5060>
[2011-11-03 06:48:03] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK84fda8269853BA89;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=79DAA10-211CA7DB
To: <sip:4122@customer.com>;tag=as71d589c0
Call-ID: f89e3598-d42ca483-7eea913a@172.30.0.197
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="185a22d9"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.197' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Reliably Transmitting (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b2f8115
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6706eaed
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71d33bcf147cc9c233f6267e317e8d2d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Really destroying SIP dialog 'cfe67e0e-c5367aa7-166259d8@172.30.0.176' Method: SUBSCRIBE
[2011-11-03 06:48:03] Retransmitting #5 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7fe241bb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKa5e293ec58365065
From: "4140" <sip:4140@customer.com>;tag=AF8644A8-D23ACD11
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 661f2035-7aecdd92-22db8323@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4140", realm="asterisk", nonce="4f3d3cd6", uri="sip:4115@customer.com:5060", response="a274ebca359170743786fa3acff3089f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (16 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:48:03] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKa5e293ec58365065;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=AF8644A8-D23ACD11
To: <sip:4115@customer.com>;tag=as18c437e1
Call-ID: 661f2035-7aecdd92-22db8323@172.30.0.175
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="140f6081"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '661f2035-7aecdd92-22db8323@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Really destroying SIP dialog '1844f365-7b9fec64-c3d42aff@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKeedd49d2BF144B4
From: "4168" <sip:4168@customer.com>;tag=3BCDF0A-9F55B46D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7182916f-eb9ad43e-b01090d1@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (15 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:48:03] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bKeedd49d2BF144B4;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=3BCDF0A-9F55B46D
To: <sip:4115@customer.com>;tag=as3a9144ae
Call-ID: 7182916f-eb9ad43e-b01090d1@172.30.0.187
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f298994"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '7182916f-eb9ad43e-b01090d1@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Retransmitting #4 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a1eff0c
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2eff3dab
To: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #11 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #2 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25a94984
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c9d92ee
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5a9aa32d3f01afce31fda86b1907ba9e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Really destroying SIP dialog 'b7988ba2-27b08275-7d13db4@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKa5d3153e3578F68F
From: "Office 113" <sip:4171@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 9708841f-46781d2c-65ddf445@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4171", realm="asterisk", nonce="69553763", uri="sip:4148@customer.com:5060", response="931cf30a6d2c3ff558c063145782280d", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (16 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4171@172.30.0.165:5060>
[2011-11-03 06:48:03] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKa5d3153e3578F68F;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>;tag=as59e1bbca
Call-ID: 9708841f-46781d2c-65ddf445@172.30.0.165
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="43f0aa66"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '9708841f-46781d2c-65ddf445@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK49e3cb04
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6e50f4
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ed9ad4b4383baeb5c442b0b014c159d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Retransmitting #8 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ec55a50
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a638217
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4a7586b4029ac9ec5d6136e87fd53b39@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a17c643
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca4208d
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 405621e02c094e0636cd9e370b156f73@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKb50ece3b778761DA
From: "4173" <sip:4173@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (15 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4173@172.30.0.178:5060>
[2011-11-03 06:48:03] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bKb50ece3b778761DA;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>;tag=as5ca45555
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.178
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="363c75be"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '78031bfd-6cc68e94-cf2353cf@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK55a1ee55
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as74380269
To: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Retransmitting #1 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4484670a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2cc38e79
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 66a6a25600f7a6fa10d867476c2585ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56aeb16c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7492b7ad
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6781810952ce9bf369c79c09674784fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:03] Retransmitting #5 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ff6416e
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7b42bfd5
To: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:03] Really destroying SIP dialog 'f4c3d4c8-d2329751-4b92475a@172.30.0.180' Method: SUBSCRIBE
[2011-11-03 06:48:03]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK31295ca018C1EDAB
From: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:03] --- (15 headers 0 lines) ---
[2011-11-03 06:48:03] Creating new subscription
[2011-11-03 06:48:03] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:48:03] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:48:03] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:48:03]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK31295ca018C1EDAB;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
To: <sip:4119@customer.com>;tag=as0585287b
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53d0c268"
Content-Length: 0


<------------>
[2011-11-03 06:48:03] Scheduling destruction of SIP dialog '9ab1d38a-dba856ed-2992dc4@172.30.0.168' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:03] Really destroying SIP dialog '4ed85f33-99e3da80-72a000c9@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:48:03] Retransmitting #6 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKa0fd777b1CF8CE18
From: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
To: <sip:4122@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="61ab4183", uri="sip:4122@customer.com:5060", response="76a4acca1f3ec0e5fed0abd16e4b21a7", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (16 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:48:04] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:48:04] Looking for 4122 in default-local (domain customer.com:5060)
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog '3f183dd3-50755910-9d8e4599@172.30.0.177' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKa0fd777b1CF8CE18;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
To: <sip:4122@customer.com>;tag=as2f99879a
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4122@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:04] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:48:04] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:48:04] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5fe8cfef
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2f99879a
To: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3a0ed75b-16468bc-154fc1f1@172.30.0.195 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 5941ms with no response
[2011-11-03 06:48:04] Really destroying SIP dialog 'b86e3dc7-5211810e-25a67e11@172.30.0.171' Method: SUBSCRIBE
[2011-11-03 06:48:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 9a3ef30e-d0c05fd5-21af9d94@172.30.0.192 for seqno 106 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 52654ms with no response
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.188:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKe452c86d786C6484
From: "4149" <sip:4149@customer.com>;tag=E15E0934-564D8E23
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a54d81f3-8acc6c7a-dfc435d9@172.30.0.188
Contact: <sip:4149@172.30.0.188:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.188:5060 (no NAT)
[2011-11-03 06:48:04] list_route: hop: <sip:4149@172.30.0.188:5060>
[2011-11-03 06:48:04] Found peer '4149' for '4149' from 172.30.0.188:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.188:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKe452c86d786C6484;received=172.30.0.188
From: "4149" <sip:4149@customer.com>;tag=E15E0934-564D8E23
To: <sip:4147@customer.com>;tag=as34ca2468
Call-ID: a54d81f3-8acc6c7a-dfc435d9@172.30.0.188
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20f3c290"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog 'a54d81f3-8acc6c7a-dfc435d9@172.30.0.188' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK1153d83a5A70009B
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:04] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK1153d83a5A70009B;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as3c290715
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a6bd63c"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog 'ddf9dab8-ab9de2a1-75ab5826@172.30.0.163' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04] Really destroying SIP dialog 'd50cf517-379c317c-661f2035@172.30.0.196' Method: SUBSCRIBE
[2011-11-03 06:48:04] Retransmitting #3 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK021bbf7b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5d198f08
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0cdf6180063048001b065f53446adef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Retransmitting #4 (no NAT) to 172.30.0.183:5060:
OPTIONS sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK36762d5b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as37e5b7bd
To: <sip:4147@172.30.0.183:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 49462d9a03fa9094643793f873b2bdef@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Really destroying SIP dialog '49462d9a03fa9094643793f873b2bdef@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:04] Retransmitting #6 (no NAT) to 172.30.0.167:5060:
NOTIFY sip:4145@172.30.0.167:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3add5ebd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as1c4480b3
To: "4145" <sip:4145@customer.com>;tag=8BFDED2E-17854BDF
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ecd301f0-4e178379-e5426f4e@172.30.0.167
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4145@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] Retransmitting #6 (no NAT) to 172.30.0.195:5060:
NOTIFY sip:4113@172.30.0.195:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK533fe5f6
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7a9dfdc2
To: "Office 131" <sip:4113@customer.com>;tag=63B2E167-E132AEB8
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 3a3db443-9199e1c4-78b99199@172.30.0.195
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4113@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKa3ba2d85FA2B9B04
From: "Office 105" <sip:4131@customer.com>;tag=BC8173AC-912FFA67
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7452d01f-f76a8dc6-903f12a9@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:48:04] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:48:04] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKa3ba2d85FA2B9B04;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=BC8173AC-912FFA67
To: <sip:4119@customer.com>;tag=as3cc3c85f
Call-ID: 7452d01f-f76a8dc6-903f12a9@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56099cbc"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog '7452d01f-f76a8dc6-903f12a9@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04] Retransmitting #5 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07707bac
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as3a063028
To: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 136b5158-e366ebe1-5d9773ea@172.30.0.180 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 9773ms with no response
[2011-11-03 06:48:04] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2205b815
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35210e4e
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as00e218e3
To: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission ecd301f0-4e178379-e5426f4e@172.30.0.167 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6413ms with no response
[2011-11-03 06:48:04] Retransmitting #4 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27283f8b
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as3f9e7318
To: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 3a3db443-9199e1c4-78b99199@172.30.0.195 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6411ms with no response
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4135@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK1780c41aFECF63C1
From: "4118" <sip:4118@customer.com>;tag=D30B1C9A-5D282441
To: <sip:4135@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a7eae618-d47d6a1f-552d3ed6@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:48:04] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:48:04] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK1780c41aFECF63C1;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=D30B1C9A-5D282441
To: <sip:4135@customer.com>;tag=as2872693e
Call-ID: a7eae618-d47d6a1f-552d3ed6@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="32b4602b"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog 'a7eae618-d47d6a1f-552d3ed6@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47a91343
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKe296db19973FD306
From: "4140" <sip:4140@customer.com>;tag=378B497B-70D4C670
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 61221967-8836730c-98ca2405@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:48:04] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:48:04] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bKe296db19973FD306;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=378B497B-70D4C670
To: <sip:4127@customer.com>;tag=as135dc89d
Call-ID: 61221967-8836730c-98ca2405@172.30.0.175
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09ccfcc4"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog '61221967-8836730c-98ca2405@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04] Retransmitting #8 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] Retransmitting #4 (no NAT) to 172.30.0.176:5060:
OPTIONS sip:4122@172.30.0.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7d5004a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4aeb60e4
To: <sip:4122@172.30.0.176:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7f78d35e27cb790e2911e7437e5d9ccb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Really destroying SIP dialog '7f78d35e27cb790e2911e7437e5d9ccb@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4116@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK6cb925ef4960FE0
From: "Office 131" <sip:4113@customer.com>;tag=54FFE683-D2A19104
To: <sip:4116@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 630510e5-2b854b8e-91a879ab@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:48:04] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:48:04] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK6cb925ef4960FE0;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=54FFE683-D2A19104
To: <sip:4116@customer.com>;tag=as51cfbde3
Call-ID: 630510e5-2b854b8e-91a879ab@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="13114bbe"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog '630510e5-2b854b8e-91a879ab@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5fe8cfef
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2f99879a
To: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] Retransmitting #1 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b2f8115
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6706eaed
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71d33bcf147cc9c233f6267e317e8d2d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK8442b2e3C1452592
From: "4124" <sip:4124@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:48:04] list_route: hop: <sip:4124@172.30.0.164:5060>
[2011-11-03 06:48:04] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK8442b2e3C1452592;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as7a4a5511
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.164
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="163fdb3a"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog 'c5f32c09-76ce4e08-4f93254f@172.30.0.164' in 51456 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa57b3ab2A86B6DD9
From: "4118" <sip:4118@customer.com>;tag=1682AE66-728347CD
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 68c5dbb0-ad042337-54b4136e@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (15 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:48:04] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:48:04] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bKa57b3ab2A86B6DD9;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=1682AE66-728347CD
To: <sip:4119@customer.com>;tag=as0bcddf8c
Call-ID: 68c5dbb0-ad042337-54b4136e@172.30.0.173
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f85c292"
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog '68c5dbb0-ad042337-54b4136e@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04] Retransmitting #3 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25a94984
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c9d92ee
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5a9aa32d3f01afce31fda86b1907ba9e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK49e3cb04
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6e50f4
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ed9ad4b4383baeb5c442b0b014c159d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ec55a50
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a638217
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4a7586b4029ac9ec5d6136e87fd53b39@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fd046b7
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKc0af2da2C316772B
From: "Office 106" <sip:4120@customer.com>;tag=42BDEE3D-C7CD8046
To: <sip:4108@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 35317d09-75c0c112-fd33329b@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="5e895934", uri="sip:4108@customer.com:5060", response="7d5f027361eb64c716f98d205e0e736f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (16 headers 0 lines) ---
[2011-11-03 06:48:04] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:04] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:48:04] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 106" <sip:4120@customer.com>;tag=42BDEE3D-C7CD8046'
[2011-11-03 06:48:04]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKc0af2da2C316772B;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=42BDEE3D-C7CD8046
To: <sip:4108@customer.com>;tag=as317802ec
Call-ID: 35317d09-75c0c112-fd33329b@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d87e8fd", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:48:04] Scheduling destruction of SIP dialog '35317d09-75c0c112-fd33329b@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:04] Retransmitting #4 (no NAT) to 172.30.0.168:5060:
OPTIONS sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4a17c643
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7ca4208d
To: <sip:4125@172.30.0.168:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 405621e02c094e0636cd9e370b156f73@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Really destroying SIP dialog '405621e02c094e0636cd9e370b156f73@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:04] Retransmitting #2 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4484670a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2cc38e79
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 66a6a25600f7a6fa10d867476c2585ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04] Retransmitting #7 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:04] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56aeb16c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7492b7ad
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6781810952ce9bf369c79c09674784fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:04]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKfc59e7546285FBDD
From: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
To: <sip:4138@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4120", realm="asterisk", nonce="218e7b27", uri="sip:4138@customer.com:5060", response="dec6a125e4dedf8033aef5394e5bcf42", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:04] --- (16 headers 0 lines) ---
[2011-11-03 06:48:04] Creating new subscription
[2011-11-03 06:48:04] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:48:04] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:48:05] Looking for 4138 in default-local (domain customer.com:5060)
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog 'fde2cac4-4612074d-b2293156@172.30.0.180' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKfc59e7546285FBDD;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
To: <sip:4138@customer.com>;tag=as0e9574c5
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4138@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:05] set_destination: Parsing <sip:4120@172.30.0.180:5060> for address/port to send to
[2011-11-03 06:48:05] set_destination: set destination to 172.30.0.180:5060
[2011-11-03 06:48:05] Reliably Transmitting (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK186417c9
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as0e9574c5
To: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Really destroying SIP dialog 'e56f2582-f4496c2d-8cab4c08@172.30.0.166' Method: SUBSCRIBE
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK8fa72154B181096D
From: "4140" <sip:4140@customer.com>;tag=5112CAB4-B941C04D
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 4740daa-6b0f1d9b-bd552310@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (15 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:48:05] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK8fa72154B181096D;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=5112CAB4-B941C04D
To: <sip:4109@customer.com>;tag=as0ff76f2c
Call-ID: 4740daa-6b0f1d9b-bd552310@172.30.0.175
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="638e7804"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog '4740daa-6b0f1d9b-bd552310@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK4e2ea093F2169854
From: "Office 131" <sip:4113@customer.com>;tag=1E83FF0A-5425C1D7
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: df1ce224-c5b03179-e464acb2@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="3bf4346d", uri="sip:4115@customer.com:5060", response="c18045ad985576bf87269b856dc439e6", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (16 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:48:05] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK4e2ea093F2169854;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=1E83FF0A-5425C1D7
To: <sip:4115@customer.com>;tag=as0c26361f
Call-ID: df1ce224-c5b03179-e464acb2@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c428ba4"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog 'df1ce224-c5b03179-e464acb2@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Retransmitting #4 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b272693
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as24d6fe6e
To: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKe8c3dcdaEEB51D8D
From: "Office 105" <sip:4131@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: cf3c3e2c-ca4cdee7-7764d2ae@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (15 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:48:05] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKe8c3dcdaEEB51D8D;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>;tag=as1ddc7292
Call-ID: cf3c3e2c-ca4cdee7-7764d2ae@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39d59ca7"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog 'cf3c3e2c-ca4cdee7-7764d2ae@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Retransmitting #1 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK186417c9
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as0e9574c5
To: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Retransmitting #4 (no NAT) to 172.30.0.165:5060:
OPTIONS sip:4171@172.30.0.165:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK021bbf7b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5d198f08
To: <sip:4171@172.30.0.165:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 0cdf6180063048001b065f53446adef7@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05] Really destroying SIP dialog '0cdf6180063048001b065f53446adef7@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:05] Really destroying SIP dialog '8b77eefa-eedd49d-2bf144b4@172.30.0.168' Method: SUBSCRIBE
[2011-11-03 06:48:05] Reliably Transmitting (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47e2b429
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2d2f2d5b
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d5167a5713a864413e11f3423f62d23@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05] Retransmitting #5 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a1eff0c
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2eff3dab
To: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc716d2fa72870C31
From: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (15 headers 0 lines) ---
[2011-11-03 06:48:05] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:05] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKc716d2fa72870C31;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as0135ac39
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6fb0812a"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Really destroying SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.199' Method: SUBSCRIBE
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4106@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKc50bab2768EBDC30
From: "Office 106" <sip:4120@customer.com>;tag=2B55645F-C3126968
To: <sip:4106@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d6b2bab9-781486c2-cd25804b@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (15 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:48:05] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bKc50bab2768EBDC30;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=2B55645F-C3126968
To: <sip:4106@customer.com>;tag=as1493c9d8
Call-ID: d6b2bab9-781486c2-cd25804b@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7b77b364"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog 'd6b2bab9-781486c2-cd25804b@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Retransmitting #8 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Retransmitting #6 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Retransmitting #4 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19aa9e2b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4116@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK3f2b15a913B0C5BE
From: "4145" <sip:4145@customer.com>;tag=FCA34E71-97556E6
To: <sip:4116@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a407c92f-5cacf8cc-9ae50d25@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (15 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:48:05] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK3f2b15a913B0C5BE;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=FCA34E71-97556E6
To: <sip:4116@customer.com>;tag=as0f0e5a3f
Call-ID: a407c92f-5cacf8cc-9ae50d25@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c3872be"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog 'a407c92f-5cacf8cc-9ae50d25@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Retransmitting #2 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK186417c9
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as0e9574c5
To: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5fe8cfef
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2f99879a
To: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Retransmitting #2 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b2f8115
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6706eaed
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71d33bcf147cc9c233f6267e317e8d2d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05] Really destroying SIP dialog '7cf1ab91-1b38956-1e32847@172.30.0.163' Method: SUBSCRIBE
[2011-11-03 06:48:05] Retransmitting #4 (no NAT) to 172.30.0.166:5060:
OPTIONS sip:4137@172.30.0.166:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK25a94984
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c9d92ee
To: <sip:4137@172.30.0.166:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5a9aa32d3f01afce31fda86b1907ba9e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05] Really destroying SIP dialog '5a9aa32d3f01afce31fda86b1907ba9e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4106@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK51e612377C4A23C8
From: "Office 131" <sip:4113@customer.com>;tag=DE280EAE-6341C34B
To: <sip:4106@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 7f8bf698-7add0bfd-1a758a86@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="1743dccb", uri="sip:4106@customer.com:5060", response="47f4ee52526cde8beade96926db9199d", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (16 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:48:05] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bK51e612377C4A23C8;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=DE280EAE-6341C34B
To: <sip:4106@customer.com>;tag=as55cd4bc4
Call-ID: 7f8bf698-7add0bfd-1a758a86@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74d3dc15"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog '7f8bf698-7add0bfd-1a758a86@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
OPTIONS sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK49e3cb04
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as3c6e50f4
To: <sip:4138@172.30.0.163:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ed9ad4b4383baeb5c442b0b014c159d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05] Really destroying SIP dialog '2ed9ad4b4383baeb5c442b0b014c159d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:05] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
OPTIONS sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1ec55a50
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0a638217
To: <sip:4115@172.30.0.197:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4a7586b4029ac9ec5d6136e87fd53b39@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05] Really destroying SIP dialog '4a7586b4029ac9ec5d6136e87fd53b39@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:05] Retransmitting #6 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK781d0a2aD7FDCDE1
From: "empty station" <sip:4119@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2843320-a76923a7-aea30b36@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (15 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:48:05] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK781d0a2aD7FDCDE1;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>;tag=as5f4c68f0
Call-ID: 2843320-a76923a7-aea30b36@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f6f7c88"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog '2843320-a76923a7-aea30b36@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK55a1ee55
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as74380269
To: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Retransmitting #3 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4484670a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2cc38e79
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 66a6a25600f7a6fa10d867476c2585ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56aeb16c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7492b7ad
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6781810952ce9bf369c79c09674784fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.175:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK8dfb4cb8B3CF1161
From: "4140" <sip:4140@customer.com>;tag=33C5AE9F-98B2FF24
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 548a606e-e68ad26f-eed45e34@172.30.0.175
Contact: <sip:4140@172.30.0.175:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (15 headers 0 lines) ---
[2011-11-03 06:48:05] Creating new subscription
[2011-11-03 06:48:05] Sending to 172.30.0.175:5060 (no NAT)
[2011-11-03 06:48:05] list_route: hop: <sip:4140@172.30.0.175:5060>
[2011-11-03 06:48:05] Found peer '4140' for '4140' from 172.30.0.175:5060
[2011-11-03 06:48:05]
<--- Transmitting (no NAT) to 172.30.0.175:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.175:5060;branch=z9hG4bK8dfb4cb8B3CF1161;received=172.30.0.175
From: "4140" <sip:4140@customer.com>;tag=33C5AE9F-98B2FF24
To: <sip:4147@customer.com>;tag=as56371eba
Call-ID: 548a606e-e68ad26f-eed45e34@172.30.0.175
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d47b2aa"
Content-Length: 0


<------------>
[2011-11-03 06:48:05] Scheduling destruction of SIP dialog '548a606e-e68ad26f-eed45e34@172.30.0.175' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:05] Really destroying SIP dialog 'e7b8eeb5-a95f4612-4fb66da3@172.30.0.175' Method: REGISTER
[2011-11-03 06:48:05] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27283f8b
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as3f9e7318
To: "4148" <sip:4148@customer.com>;tag=F565DE62-DC09C30D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 312cc0b9-ab7a3354-7ac3461f@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:05] Really destroying SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.170' Method: REGISTER
[2011-11-03 06:48:05]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK541f7328D3AB642F
From: "4118" <sip:4118@customer.com>;tag=1BD55451-10F09BA8
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 85339b5d-941df874-9f68d33b@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="4249893a", uri="sip:4147@customer.com:5060", response="921e738942b6187975632e3f7c6dc18e", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:05] --- (16 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4118@172.30.0.173:5060>
[2011-11-03 06:48:06] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK541f7328D3AB642F;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=1BD55451-10F09BA8
To: <sip:4147@customer.com>;tag=as41fac7a2
Call-ID: 85339b5d-941df874-9f68d33b@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66373091"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog '85339b5d-941df874-9f68d33b@172.30.0.173' in 10688 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4108@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK8d759981A61D4208
From: "4173" <sip:4173@customer.com>;tag=E07B4379-52A6EC20
To: <sip:4108@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: ea78b573-5bfe1432-53f9ccf5@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (15 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4173@172.30.0.178:5060>
[2011-11-03 06:48:06] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK8d759981A61D4208;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=E07B4379-52A6EC20
To: <sip:4108@customer.com>;tag=as0c5a81a5
Call-ID: ea78b573-5bfe1432-53f9ccf5@172.30.0.178
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7fa102d5"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog 'ea78b573-5bfe1432-53f9ccf5@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06] Reliably Transmitting (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6c55e8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as31687a6a
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c286d7722b24420669a05493094337b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:06] Retransmitting #3 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK186417c9
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as0e9574c5
To: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bKc5367aa7166259D8
From: "Office 108" <sip:4128@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 6060cfe1-e09e0702-2ff02f7b@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (15 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.182:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4128@172.30.0.182:5060>
[2011-11-03 06:48:06] Found peer '4128' for '4128' from 172.30.0.182:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bKc5367aa7166259D8;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=65FA08C7-CCA82F8
To: <sip:4127@customer.com>;tag=as0c4d8783
Call-ID: 6060cfe1-e09e0702-2ff02f7b@172.30.0.182
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="499058ce"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog '6060cfe1-e09e0702-2ff02f7b@172.30.0.182' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKfd35048aC51A9557
From: "Office 131" <sip:4113@customer.com>;tag=5120D6D4-E7962FE9
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 1482d68-8deb7f0d-2e26fed6@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (15 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:48:06] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKfd35048aC51A9557;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=5120D6D4-E7962FE9
To: <sip:4140@customer.com>;tag=as7fb737c7
Call-ID: 1482d68-8deb7f0d-2e26fed6@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3aed310d"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog '1482d68-8deb7f0d-2e26fed6@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06] Reliably Transmitting (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12da038b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as11d913e0
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2d74748b177a1f29295abd52538bb0eb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:06] Retransmitting #7 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4122@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK22da1e6aA2BB164D
From: "4173" <sip:4173@customer.com>;tag=67D2CA3C-121DD8D7
To: <sip:4122@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: aa9e0fa4-bf6e809f-4b7016ae@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (15 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4173@172.30.0.178:5060>
[2011-11-03 06:48:06] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK22da1e6aA2BB164D;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=67D2CA3C-121DD8D7
To: <sip:4122@customer.com>;tag=as0103a9ad
Call-ID: aa9e0fa4-bf6e809f-4b7016ae@172.30.0.178
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63022ab8"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog 'aa9e0fa4-bf6e809f-4b7016ae@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06] Retransmitting #1 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47e2b429
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2d2f2d5b
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d5167a5713a864413e11f3423f62d23@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:06] Really destroying SIP dialog 'b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182' Method: SUBSCRIBE
[2011-11-03 06:48:06] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78841311
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK1964fe6f37F89A60
From: "Office 108" <sip:4128@customer.com>;tag=CCEF9FC3-1FFD0894
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4128", realm="asterisk", nonce="27dbd119", uri="sip:4147@customer.com:5060", response="a962dc6a387277c558b3e977043b51ea", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (16 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.182:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4128@172.30.0.182:5060>
[2011-11-03 06:48:06] Found peer '4128' for '4128' from 172.30.0.182:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK1964fe6f37F89A60;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=CCEF9FC3-1FFD0894
To: <sip:4147@customer.com>;tag=as7e6c4665
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fcffb38"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog 'b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06] Retransmitting #5 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54133146
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06] Really destroying SIP dialog 'f41d8a29-20eba810-5722db5b@172.30.0.178' Method: REGISTER
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK5095fd1321561D52
From: "4125" <sip:4125@customer.com>;tag=369F5F15-B84CCD4C
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: df821195-814035cc-2baac627@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (15 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4125@172.30.0.168:5060>
[2011-11-03 06:48:06] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK5095fd1321561D52;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=369F5F15-B84CCD4C
To: <sip:4140@customer.com>;tag=as0b681e42
Call-ID: df821195-814035cc-2baac627@172.30.0.168
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b5e58ae"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog 'df821195-814035cc-2baac627@172.30.0.168' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06] Retransmitting #7 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK64fdddd2
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as44540955
To: "empty station" <sip:4119@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 5e4c9262-197fee59-47d4d9d8@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK6bf3e33d30250654
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="0dcdf702", uri="sip:4138@customer.com:5060", response="5c030d1cb7f7882c470b0aa77f099186", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (16 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:48:06] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:48:06] Looking for 4138 in default-local (domain customer.com:5060)
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog '31caffa2-d80e65c9-ba5c76a0@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK6bf3e33d30250654;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>;tag=as1e18c6e3
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4138@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:06] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:48:06] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:48:06] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f8a6b1a
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06] Retransmitting #3 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b2f8115
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6706eaed
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71d33bcf147cc9c233f6267e317e8d2d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:06] Really destroying SIP dialog 'a756ff0f-26f3e83e-55482e21@172.30.0.189' Method: SUBSCRIBE
[2011-11-03 06:48:06] Retransmitting #4 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dc953f5
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as0a17f176
To: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06] Really destroying SIP dialog '3ae67647-347c00d6-458f082d@172.30.0.191' Method: SUBSCRIBE
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4106@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKf5dfea84B16D47D
From: "4145" <sip:4145@customer.com>;tag=7FCEC930-D3EEF9B9
To: <sip:4106@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 14bc7002-43b06a23-9275fab0@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (15 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:48:06] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bKf5dfea84B16D47D;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=7FCEC930-D3EEF9B9
To: <sip:4106@customer.com>;tag=as58a0edca
Call-ID: 14bc7002-43b06a23-9275fab0@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61cefc05"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog '14bc7002-43b06a23-9275fab0@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK088093cd
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5371e3da
To: "Office 106" <sip:4120@customer.com>;tag=B9F5F2D1-5CEE2DA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: a6926e31-5b106e3a-a40b9bc3@172.30.0.180
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.181:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK1c0e3ca43C45281
From: "Office 107" <sip:4141@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 77964ac0-3ae67647-347c00d6@172.30.0.181
Contact: <sip:4141@172.30.0.181:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (15 headers 0 lines) ---
[2011-11-03 06:48:06] Creating new subscription
[2011-11-03 06:48:06] Sending to 172.30.0.181:5060 (no NAT)
[2011-11-03 06:48:06] list_route: hop: <sip:4141@172.30.0.181:5060>
[2011-11-03 06:48:06] Found peer '4141' for '4141' from 172.30.0.181:5060
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.181:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.181:5060;branch=z9hG4bK1c0e3ca43C45281;received=172.30.0.181
From: "Office 107" <sip:4141@customer.com>;tag=54F74FCF-2A06041E
To: <sip:4127@customer.com>;tag=as7434f8db
Call-ID: 77964ac0-3ae67647-347c00d6@172.30.0.181
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e8a59e5"
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog '77964ac0-3ae67647-347c00d6@172.30.0.181' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:06] Really destroying SIP dialog '8fcb19f5-56200334-868f8b0f@172.30.0.197' Method: REGISTER
[2011-11-03 06:48:06] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f95cdd5
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06] Retransmitting #4 (no NAT) to 172.30.0.184:5060:
OPTIONS sip:4123@172.30.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4484670a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2cc38e79
To: <sip:4123@172.30.0.184:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 66a6a25600f7a6fa10d867476c2585ed@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:06] Really destroying SIP dialog '66a6a25600f7a6fa10d867476c2585ed@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:06] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f8a6b1a
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:06]
<--- SIP read from UDP:172.30.0.177:5060 --->
REGISTER sip:customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKa5d3153e3578F68F
From: "4135" <sip:4135@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4135@customer.com>
CSeq: 2 REGISTER
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4135", realm="asterisk", nonce="51918e05", uri="sip:customer.com:5060", response="676e2306c1103ee818d84d9a23421d9a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:06] --- (13 headers 0 lines) ---
[2011-11-03 06:48:06] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:48:06]     -- Registered SIP '4135' at 172.30.0.177:5060
[2011-11-03 06:48:06] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
OPTIONS sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6d6e7517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fbefaa9
To: <sip:4135@172.30.0.177:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 786c9a7415b8cda21c9a8a240572459e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:06]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKa5d3153e3578F68F;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=32E0DE2D-1A010C82
To: <sip:4135@customer.com>;tag=as19312c1e
Call-ID: cbcecd09-935c2aee-eccb17f@172.30.0.177
CSeq: 2 REGISTER
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4135@172.30.0.177:5060>;expires=3600
Date: Thu, 03 Nov 2011 13:48:06 GMT
Content-Length: 0


<------------>
[2011-11-03 06:48:06] Scheduling destruction of SIP dialog 'cbcecd09-935c2aee-eccb17f@172.30.0.177' in 32000 ms (Method: REGISTER)
[2011-11-03 06:48:06] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
OPTIONS sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK56aeb16c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7492b7ad
To: <sip:4173@172.30.0.178:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6781810952ce9bf369c79c09674784fe@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:06]   == Extension Changed 4135[default-local] new state Idle for Notify User 4145 (queued)
[2011-11-03 06:48:06]   == Extension Changed 4135[default-local] new state Idle for Notify User 4118 (queued)
[2011-11-03 06:48:06]   == Extension Changed 4135[default-local] new state Idle for Notify User 4140 (queued)
[2011-11-03 06:48:06]   == Extension Changed 4135[default-local] new state Idle for Notify User 4124 (queued)
[2011-11-03 06:48:06]   == Extension Changed 4135[default-local] new state Idle for Notify User 4120 (queued)
[2011-11-03 06:48:06]   == Extension Changed 4135[default-local] new state Idle for Notify User 4122 (queued)
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4168 (queued)
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4138 (queued)
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4171 (queued)
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4149 (queued)
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4123 (queued)
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4134 (queued)
[2011-11-03 06:48:07] set_destination: Parsing <sip:4136@172.30.0.172:5060> for address/port to send to
[2011-11-03 06:48:07] set_destination: set destination to 172.30.0.172:5060
[2011-11-03 06:48:07] Reliably Transmitting (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09270bb1
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4136
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4173 (queued)
[2011-11-03 06:48:07] set_destination: Parsing <sip:4173@172.30.0.178:5060> for address/port to send to
[2011-11-03 06:48:07] set_destination: set destination to 172.30.0.178:5060
[2011-11-03 06:48:07] Reliably Transmitting (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59343336
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07]   == Extension Changed 4135[default-local] new state Idle for Notify User 4173
[2011-11-03 06:48:07] Really destroying SIP dialog '6781810952ce9bf369c79c09674784fe@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:07] Retransmitting #3 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07057c98
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as397e6250
To: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Really destroying SIP dialog '8c28ad86-c4c5a271-4bc8f48c@172.30.0.166' Method: SUBSCRIBE
[2011-11-03 06:48:07] Retransmitting #6 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Reliably Transmitting (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37f2dc98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4c398fc2
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ec7297323e9356605f6584e502685de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:07] Retransmitting #7 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Really destroying SIP dialog 'bdf5d4b5-60ff2a54-bae9903b@172.30.0.181' Method: SUBSCRIBE
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK2edb88703FA6AF79
From: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="2d6f9db2", uri="sip:4127@customer.com:5060", response="aeb6996ed0a82fb09080a7bc511746f9", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (16 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:48:07] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:48:07] Looking for 4127 in default-local (domain customer.com:5060)
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '75ab5826-6d5f5fd7-bf46e004@172.30.0.177' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bK2edb88703FA6AF79;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
To: <sip:4127@customer.com>;tag=as440eeb94
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4127@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:07] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:48:07] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:48:07] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4074131d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as440eeb94
To: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission e47da7d6-f2289e09-ce590c70@172.30.0.168 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6404ms with no response
[2011-11-03 06:48:07] Retransmitting #1 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6c55e8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as31687a6a
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c286d7722b24420669a05493094337b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:07] Retransmitting #6 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK23ed3909
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as551a6ae9
To: "4118" <sip:4118@customer.com>;tag=302584C3-720A92BA
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: b2085f85-8168a51c-caacd063@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #1 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59343336
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #5 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9261d0
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as252ea80a
To: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #14 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f8a6b1a
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKde119ace6397C65
From: "empty station" <sip:4119@customer.com>;tag=E4E36FDA-FE1F2A11
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3a993884-36cb0eb-977f895a@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (15 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:48:07] list_route: hop: <sip:4119@172.30.0.191:5060>
[2011-11-03 06:48:07] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKde119ace6397C65;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=E4E36FDA-FE1F2A11
To: <sip:4138@customer.com>;tag=as0b68b564
Call-ID: 3a993884-36cb0eb-977f895a@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ef00309"
Content-Length: 0


<------------>
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '3a993884-36cb0eb-977f895a@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07] Really destroying SIP dialog 'c1452592-c5f32c09-76ce4e08@172.30.0.181' Method: SUBSCRIBE
[2011-11-03 06:48:07] Retransmitting #2 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47e2b429
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2d2f2d5b
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d5167a5713a864413e11f3423f62d23@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:07] Retransmitting #1 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12da038b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as11d913e0
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2d74748b177a1f29295abd52538bb0eb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4ff6416e
From: <sip:4148@customer.com>;tag=as7b42bfd5
To: "4125" <sip:4125@customer.com>;tag=AF6D521D-30E74C34
CSeq: 102 NOTIFY
Call-ID: e47da7d6-f2289e09-ce590c70@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (11 headers 0 lines) ---
[2011-11-03 06:48:07] Retransmitting #4 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK186417c9
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as0e9574c5
To: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #2 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59343336
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK59c9257fB7400EF0
From: "Office 108" <sip:4128@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (15 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.182:5060 (no NAT)
[2011-11-03 06:48:07] list_route: hop: <sip:4128@172.30.0.182:5060>
[2011-11-03 06:48:07] Found peer '4128' for '4128' from 172.30.0.182:5060
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK59c9257fB7400EF0;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=9981B6C4-710945ED
To: <sip:4109@customer.com>;tag=as670d9618
Call-ID: 7fd09539-8cc4f29a-6f06e753@172.30.0.182
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16d08941"
Content-Length: 0


<------------>
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '7fd09539-8cc4f29a-6f06e753@172.30.0.182' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4130@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK8efbcbc6F2B45E97
From: "4145" <sip:4145@customer.com>;tag=EFEA983E-470FA9AF
To: <sip:4130@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 484b4a14-f56e4ccd-51e6d212@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (15 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:48:07] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:48:07] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK8efbcbc6F2B45E97;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=EFEA983E-470FA9AF
To: <sip:4130@customer.com>;tag=as4652e39d
Call-ID: 484b4a14-f56e4ccd-51e6d212@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15c4e325"
Content-Length: 0


<------------>
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '484b4a14-f56e4ccd-51e6d212@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07] Really destroying SIP dialog 'e8fa9348-b82111f1-f5635466@172.30.0.167' Method: SUBSCRIBE
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK613a66d6C3D2A877
From: "4108" <sip:4108@customer.com>;tag=AEAE56DC-26394315
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 5dd09a5c-b132b495-3cc06b72@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (15 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:48:07] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:48:07] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK613a66d6C3D2A877;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=AEAE56DC-26394315
To: <sip:4147@customer.com>;tag=as4a84d73e
Call-ID: 5dd09a5c-b132b495-3cc06b72@172.30.0.196
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="335af351"
Content-Length: 0


<------------>
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '5dd09a5c-b132b495-3cc06b72@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07] Retransmitting #1 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09270bb1
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ae4517c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a3eacbf
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 172b465e7557c12463787e9b1fe120b1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKde119ace6397C65
From: "Leon Zhong" <sip:4134@customer.com>;tag=E4E36FDA-FE1F2A11
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3a993884-36cb0eb-977f895a@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (15 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:48:07] list_route: hop: <sip:4134@172.30.0.179:5060>
[2011-11-03 06:48:07] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bKde119ace6397C65;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=E4E36FDA-FE1F2A11
To: <sip:4138@customer.com>;tag=as266f4972
Call-ID: 3a993884-36cb0eb-977f895a@172.30.0.179
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15208473"
Content-Length: 0


<------------>
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '3a993884-36cb0eb-977f895a@172.30.0.179' in 98944 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07] Retransmitting #6 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7fe241bb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5fe8cfef
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2f99879a
To: "4135" <sip:4135@customer.com>;tag=821408B2-3FC83533
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 3f183dd3-50755910-9d8e4599@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #4 (no NAT) to 172.30.0.188:5060:
OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7b2f8115
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as6706eaed
To: <sip:4149@172.30.0.188:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 71d33bcf147cc9c233f6267e317e8d2d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:07] Really destroying SIP dialog '71d33bcf147cc9c233f6267e317e8d2d@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4135@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK92a4d0a1B8DB293
From: "Office 106" <sip:4120@customer.com>;tag=D82EBBFA-79568983
To: <sip:4135@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2136859c-69f90625-8c09742e@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (15 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.180:5060 (no NAT)
[2011-11-03 06:48:07] list_route: hop: <sip:4120@172.30.0.180:5060>
[2011-11-03 06:48:07] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK92a4d0a1B8DB293;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=D82EBBFA-79568983
To: <sip:4135@customer.com>;tag=as205f0fd6
Call-ID: 2136859c-69f90625-8c09742e@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="056dc74f"
Content-Length: 0


<------------>
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '2136859c-69f90625-8c09742e@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4074131d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as440eeb94
To: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a6926e31-5b106e3a-a40b9bc3@172.30.0.180 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 9769ms with no response
[2011-11-03 06:48:07] Retransmitting #12 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6606018d
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as52d83bb4
To: "4173" <sip:4173@customer.com>;tag=9C84FEF8-2C65DB8F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: f209ee5a-b8b6bc81-bf235ea0@172.30.0.178
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK75dcdcb771194164
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="2a6bd63c", uri="sip:4109@customer.com:5060", response="f2a0fbc883a729393d2a0c0410888e80", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (16 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.163:5060 (no NAT)
[2011-11-03 06:48:07] Found peer '4138' for '4138' from 172.30.0.163:5060
[2011-11-03 06:48:07] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog 'ddf9dab8-ab9de2a1-75ab5826@172.30.0.163' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK75dcdcb771194164;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as3c290715
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:07] set_destination: Parsing <sip:4138@172.30.0.163:5060> for address/port to send to
[2011-11-03 06:48:07] set_destination: set destination to 172.30.0.163:5060
[2011-11-03 06:48:07] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK434b6e17
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #9 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK486e9c73
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as38a8c95c
To: "4124" <sip:4124@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 34d99d55-c99f65f4-fe46e6db@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07] Retransmitting #3 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59343336
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK55a1ee55
From: <sip:4109@customer.com>;tag=as74380269
To: "4135" <sip:4135@customer.com>;tag=8FD0A098-469A901
CSeq: 102 NOTIFY
Call-ID: 65ddf445-1153d83a-5a70009b@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (11 headers 0 lines) ---
[2011-11-03 06:48:07] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:48:07] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f8a6b1a
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:07]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKaa069f8f94DC84F6
From: "Office 105" <sip:4131@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 4c6094f3-d6a78d6a-2d08a65d@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4131", realm="asterisk", nonce="1a25c239", uri="sip:4109@customer.com:5060", response="7147290e21bc1873bfdaa8feecc3bd9f", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:07] --- (16 headers 0 lines) ---
[2011-11-03 06:48:07] Creating new subscription
[2011-11-03 06:48:07] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:48:07] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:48:07] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:48:07]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bKaa069f8f94DC84F6;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=F0E9528A-93312BFD
To: <sip:4109@customer.com>;tag=as2c4919ec
Call-ID: 4c6094f3-d6a78d6a-2d08a65d@172.30.0.170
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c12ae1f"
Content-Length: 0


<------------>
[2011-11-03 06:48:07] Scheduling destruction of SIP dialog '4c6094f3-d6a78d6a-2d08a65d@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:07] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK434b6e17
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.177:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKc33531dc87A7235
From: "4135" <sip:4135@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 4c393b60-e9b9a2a9-b618f20e@172.30.0.177
Contact: <sip:4135@172.30.0.177:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4135", realm="asterisk", nonce="779f6eac", uri="sip:4148@customer.com:5060", response="bb006773f80cf5734263bab144fc2171", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (16 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.177:5060 (no NAT)
[2011-11-03 06:48:08] Found peer '4135' for '4135' from 172.30.0.177:5060
[2011-11-03 06:48:08] Looking for 4148 in default-local (domain customer.com:5060)
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog '4c393b60-e9b9a2a9-b618f20e@172.30.0.177' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.177:5060;branch=z9hG4bKc33531dc87A7235;received=172.30.0.177
From: "4135" <sip:4135@customer.com>;tag=3A0EBCA5-24D5B71A
To: <sip:4148@customer.com>;tag=as26cc82d5
Call-ID: 4c393b60-e9b9a2a9-b618f20e@172.30.0.177
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4148@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:08] set_destination: Parsing <sip:4135@172.30.0.177:5060> for address/port to send to
[2011-11-03 06:48:08] set_destination: set destination to 172.30.0.177:5060
[2011-11-03 06:48:08] Reliably Transmitting (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK40f2adbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as26cc82d5
To: "4135" <sip:4135@customer.com>;tag=3A0EBCA5-24D5B71A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 4c393b60-e9b9a2a9-b618f20e@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
OPTIONS sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6d6e7517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fbefaa9
To: <sip:4135@172.30.0.177:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 786c9a7415b8cda21c9a8a240572459e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:08] Retransmitting #7 (no NAT) to 172.30.0.183:5060:
NOTIFY sip:4147@172.30.0.183:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK53131414
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as17dabd16
To: "4147" <sip:4147@customer.com>;tag=65FA08C7-CCA82F8
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: cfe67e0e-c5367aa7-166259d8@172.30.0.183
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4147@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKacd8168675DCDCB7
From: "Office 113" <sip:4171@customer.com>;tag=4585B9BC-9BF7295
To: <sip:4115@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 71194164-43057d9d-996bda32@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (15 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:48:08] list_route: hop: <sip:4171@172.30.0.165:5060>
[2011-11-03 06:48:08] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bKacd8168675DCDCB7;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=4585B9BC-9BF7295
To: <sip:4115@customer.com>;tag=as08c453e4
Call-ID: 71194164-43057d9d-996bda32@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0227fd71"
Content-Length: 0


<------------>
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog '71194164-43057d9d-996bda32@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08] Retransmitting #1 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37f2dc98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4c398fc2
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ec7297323e9356605f6584e502685de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:08] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK434b6e17
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #2 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6c55e8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as31687a6a
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c286d7722b24420669a05493094337b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK6bf3e33d30250654
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="0dcdf702", uri="sip:4138@customer.com:5060", response="5c030d1cb7f7882c470b0aa77f099186", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (16 headers 0 lines) ---
[2011-11-03 06:48:08] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:08] Looking for 4138 in default-local (domain customer.com:5060)
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog '31caffa2-d80e65c9-ba5c76a0@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK6bf3e33d30250654;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
To: <sip:4138@customer.com>;tag=as1e18c6e3
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4138@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:08] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:48:08] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:48:08] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c103a38
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.167:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK3a5992e8A73B911
From: "4145" <sip:4145@customer.com>;tag=375B3CEC-EB1A1EC5
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: bdc80d06-e8cc6d7-e963cd54@172.30.0.167
Contact: <sip:4145@172.30.0.167:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (15 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.167:5060 (no NAT)
[2011-11-03 06:48:08] list_route: hop: <sip:4145@172.30.0.167:5060>
[2011-11-03 06:48:08] Found peer '4145' for '4145' from 172.30.0.167:5060
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.167:5060;branch=z9hG4bK3a5992e8A73B911;received=172.30.0.167
From: "4145" <sip:4145@customer.com>;tag=375B3CEC-EB1A1EC5
To: <sip:4147@customer.com>;tag=as0c39163a
Call-ID: bdc80d06-e8cc6d7-e963cd54@172.30.0.167
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39b036db"
Content-Length: 0


<------------>
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog 'bdc80d06-e8cc6d7-e963cd54@172.30.0.167' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08] Reliably Transmitting (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20a3af7c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as498bf214
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60f0a912692004145d32b44a156d546d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:08] Retransmitting #6 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07707bac
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as3a063028
To: "empty station" <sip:4119@customer.com>;tag=A358A061-CD1751A0
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: db6f3505-b4cfe424-a9e0f78b@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #3 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47e2b429
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2d2f2d5b
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d5167a5713a864413e11f3423f62d23@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:08] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2205b815
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as43be8f18
To: "4116" <sip:4116@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.199
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #2 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12da038b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as11d913e0
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2d74748b177a1f29295abd52538bb0eb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:08] Retransmitting #6 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK35210e4e
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as00e218e3
To: "Office 106" <sip:4120@customer.com>;tag=413B81A9-E2E5B5B2
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: a088c697-bbc01fa0-e89a2629@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.164:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK65c5d3505E18C657
From: "4124" <sip:4124@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: a76923a7-aea30b36-e90f478d@172.30.0.164
Contact: <sip:4124@172.30.0.164:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4124", realm="asterisk", nonce="12b90c0a", uri="sip:4119@customer.com:5060", response="b069b169b8316690bdc8cc676a791d61", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (16 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.164:5060 (no NAT)
[2011-11-03 06:48:08] Found peer '4124' for '4124' from 172.30.0.164:5060
[2011-11-03 06:48:08] Looking for 4119 in default-local (domain customer.com:5060)
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog 'a76923a7-aea30b36-e90f478d@172.30.0.164' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.164:5060;branch=z9hG4bK65c5d3505E18C657;received=172.30.0.164
From: "4124" <sip:4124@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>;tag=as64914f5d
Call-ID: a76923a7-aea30b36-e90f478d@172.30.0.164
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4119@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:08] set_destination: Parsing <sip:4124@172.30.0.164:5060> for address/port to send to
[2011-11-03 06:48:08] set_destination: set destination to 172.30.0.164:5060
[2011-11-03 06:48:08] Reliably Transmitting (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ac0e69a
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as64914f5d
To: "4124" <sip:4124@customer.com>;tag=30F788B8-C2E26E7F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: a76923a7-aea30b36-e90f478d@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #6 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47a91343
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as7574ce02
To: "4136" <sip:4136@customer.com>;tag=E73FAEBB-CBC17AD4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: f6474834-25214b0d-6a9d0ca6@172.30.0.172
CSeq: 108 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Really destroying SIP dialog '43c45281-77964ac0-3ae67647@172.30.0.199' Method: SUBSCRIBE
[2011-11-03 06:48:08] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c103a38
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4115@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKda74979165C5D350
From: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4119", realm="asterisk", nonce="6fb0812a", uri="sip:4115@customer.com:5060", response="04406bfaf069124ae9cbb131a93517af", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (16 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.191:5060 (no NAT)
[2011-11-03 06:48:08] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:48:08] Looking for 4115 in default-local (domain customer.com:5060)
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog '6a185af0-a88708f7-7ef9e206@172.30.0.191' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bKda74979165C5D350;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
To: <sip:4115@customer.com>;tag=as0135ac39
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4115@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:08] set_destination: Parsing <sip:4119@172.30.0.191:5060> for address/port to send to
[2011-11-03 06:48:08] set_destination: set destination to 172.30.0.191:5060
[2011-11-03 06:48:08] Reliably Transmitting (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK752f0e1f
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as0135ac39
To: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #6 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6a1eff0c
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as2eff3dab
To: "4116" <sip:4116@customer.com>;tag=5FC2D8D4-741BAABB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: 8eca8c0c-d712de33-5e4c9262@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #9 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK14d365c4
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK434b6e17
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK40d1bcaf5A84C296
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="5dbddde7", uri="sip:4140@customer.com:5060", response="e5c1f11d3e6eff428d91b7e39faa4e4a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (16 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.197:5060 (no NAT)
[2011-11-03 06:48:08] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-11-03 06:48:08] Looking for 4140 in default-local (domain customer.com:5060)
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog 'f76a8dc6-903f12a9-9987ba38@172.30.0.197' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK40d1bcaf5A84C296;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>;tag=as756b08e3
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4140@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:08] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:48:08] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:48:08] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aeba898
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 8eca8c0c-d712de33-5e4c9262@172.30.0.199 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6434ms with no response
[2011-11-03 06:48:08] Retransmitting #2 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09270bb1
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #4 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59343336
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #1 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK40f2adbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as26cc82d5
To: "4135" <sip:4135@customer.com>;tag=3A0EBCA5-24D5B71A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 4c393b60-e9b9a2a9-b618f20e@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #1 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ae4517c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a3eacbf
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 172b465e7557c12463787e9b1fe120b1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:08] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission b2085f85-8168a51c-caacd063@172.30.0.173 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10650ms with no response
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4136@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc41177734DE170B4
From: "Office 131" <sip:4113@customer.com>;tag=3C349D18-BE1E547D
To: <sip:4136@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 13d339de-a62eda3b-c4eedd1c@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4113", realm="asterisk", nonce="01cd93c0", uri="sip:4136@customer.com:5060", response="1477054ba3b10839d3daec144a09139d", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (16 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:48:08] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:48:08] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc41177734DE170B4;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=3C349D18-BE1E547D
To: <sip:4136@customer.com>;tag=as668ef47c
Call-ID: 13d339de-a62eda3b-c4eedd1c@172.30.0.195
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1eb1fe82"
Content-Length: 0


<------------>
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog '13d339de-a62eda3b-c4eedd1c@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08] Really destroying SIP dialog '8d474b1d-d46049ea-53ee6bb@172.30.0.184' Method: REGISTER
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK18c1edab9AB1D38A
From: "4168" <sip:4168@customer.com>;tag=855409A8-ED043E93
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: dba856ed-2992dc4-62d4863f@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (15 headers 0 lines) ---
[2011-11-03 06:48:08] Creating new subscription
[2011-11-03 06:48:08] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:48:08] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:48:08] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK18c1edab9AB1D38A;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=855409A8-ED043E93
To: <sip:4119@customer.com>;tag=as3c425850
Call-ID: dba856ed-2992dc4-62d4863f@172.30.0.187
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4081d69a"
Content-Length: 0


<------------>
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog 'dba856ed-2992dc4-62d4863f@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4074131d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as440eeb94
To: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aeba898
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #7 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4101e130
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as2edb01b6
To: "Office 106" <sip:4120@customer.com>;tag=3F1942B5-66A56F74
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 80914c21-4966b340-c96dfc17@172.30.0.180
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c103a38
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.177:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4afc64b5
From: <sip:4109@customer.com>;tag=as20a320b3
To: "4135" <sip:4135@customer.com>;tag=BD9B8714-EF6E265D
CSeq: 106 NOTIFY
Call-ID: d73b83a7-5412e034-99a050fd@172.30.0.177
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (10 headers 0 lines) ---
[2011-11-03 06:48:08] SIP Response message for INCOMING dialog NOTIFY arrived
[2011-11-03 06:48:08] Retransmitting #3 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4fd046b7
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2ff9da69
To: "4106" <sip:4106@customer.com>;tag=A0315819-68B23F68
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 36b2a714-5b62726f-18b0f756@172.30.0.171
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #1 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK752f0e1f
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as0135ac39
To: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Really destroying SIP dialog 'd73b83a7-5412e034-99a050fd@172.30.0.177' Method: SUBSCRIBE
[2011-11-03 06:48:08]
<--- SIP read from UDP:172.30.0.182:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK1964fe6f37F89A60
From: "Office 108" <sip:4128@customer.com>;tag=CCEF9FC3-1FFD0894
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182
Contact: <sip:4128@172.30.0.182:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4128", realm="asterisk", nonce="27dbd119", uri="sip:4147@customer.com:5060", response="a962dc6a387277c558b3e977043b51ea", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:08] --- (16 headers 0 lines) ---
[2011-11-03 06:48:08] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:08] Found peer '4128' for '4128' from 172.30.0.182:5060
[2011-11-03 06:48:08] NOTICE[9661]: chan_sip.c:13994 check_auth: Correct auth, but based on stale nonce received from '"Office 108" <sip:4128@customer.com>;tag=CCEF9FC3-1FFD0894'
[2011-11-03 06:48:08]
<--- Transmitting (no NAT) to 172.30.0.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.182:5060;branch=z9hG4bK1964fe6f37F89A60;received=172.30.0.182
From: "Office 108" <sip:4128@customer.com>;tag=CCEF9FC3-1FFD0894
To: <sip:4147@customer.com>;tag=as7e6c4665
Call-ID: b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fcffb38", stale=true
Content-Length: 0


<------------>
[2011-11-03 06:48:08] Scheduling destruction of SIP dialog 'b54ec0c5-a46dfe46-9fdf9e1f@172.30.0.182' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:08] Retransmitting #8 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK524fef86
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as603feaf0
To: "4124" <sip:4124@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aeba898
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:08] Reliably Transmitting (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcb002e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as491ccb3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6045d38a5732d0883b2f11175ee5bd8e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.191:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK781d0a2aD7FDCDE1
From: "empty station" <sip:4119@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2843320-a76923a7-aea30b36@172.30.0.191
Contact: <sip:4119@172.30.0.191:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (15 headers 0 lines) ---
[2011-11-03 06:48:09] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:09] Found peer '4119' for '4119' from 172.30.0.191:5060
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.191:5060;branch=z9hG4bK781d0a2aD7FDCDE1;received=172.30.0.191
From: "empty station" <sip:4119@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>;tag=as5f4c68f0
Call-ID: 2843320-a76923a7-aea30b36@172.30.0.191
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f6f7c88"
Content-Length: 0


<------------>
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog '2843320-a76923a7-aea30b36@172.30.0.191' in 19072 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09] Really destroying SIP dialog '18535f23-983c4e92-8c852f31@172.30.0.190' Method: SUBSCRIBE
[2011-11-03 06:48:09] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
OPTIONS sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6d6e7517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fbefaa9
To: <sip:4135@172.30.0.177:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 786c9a7415b8cda21c9a8a240572459e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09] Reliably Transmitting (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3ccc2a26
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0fe734da
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7bf42ed227c7867a242679b1165a8445@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09] Reliably Transmitting (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e78a517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c6afcaa
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28965d7e4bb4d12118d538437bc60dcb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09] Retransmitting #2 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37f2dc98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4c398fc2
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ec7297323e9356605f6584e502685de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK40d1bcaf5A84C296
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="5dbddde7", uri="sip:4140@customer.com:5060", response="e5c1f11d3e6eff428d91b7e39faa4e4a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (16 headers 0 lines) ---
[2011-11-03 06:48:09] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:09] Looking for 4140 in default-local (domain customer.com:5060)
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog 'f76a8dc6-903f12a9-9987ba38@172.30.0.197' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK40d1bcaf5A84C296;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>;tag=as756b08e3
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4140@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:09] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:48:09] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:48:09] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK754b4acf
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Really destroying SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:48:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 5e4c9262-197fee59-47d4d9d8@172.30.0.191 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 19344ms with no response
[2011-11-03 06:48:09] Retransmitting #3 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6c55e8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as31687a6a
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c286d7722b24420669a05493094337b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 312cc0b9-ab7a3354-7ac3461f@172.30.0.192 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6605ms with no response
[2011-11-03 06:48:09] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7f8a6b1a
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #5 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b272693
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as24d6fe6e
To: "empty station" <sip:4119@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #1 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ac0e69a
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as64914f5d
To: "4124" <sip:4124@customer.com>;tag=30F788B8-C2E26E7F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: a76923a7-aea30b36-e90f478d@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aeba898
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #4 (no NAT) to 172.30.0.196:5060:
OPTIONS sip:4108@172.30.0.196:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK47e2b429
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as2d2f2d5b
To: <sip:4108@172.30.0.196:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 4d5167a5713a864413e11f3423f62d23@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09] Really destroying SIP dialog '4d5167a5713a864413e11f3423f62d23@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:09] Retransmitting #3 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12da038b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as11d913e0
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2d74748b177a1f29295abd52538bb0eb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09] Retransmitting #4 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK434b6e17
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission a088c697-bbc01fa0-e89a2629@172.30.0.180 for seqno 102 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 9894ms with no response
[2011-11-03 06:48:09] Retransmitting #1 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20a3af7c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as498bf214
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60f0a912692004145d32b44a156d546d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09] Really destroying SIP dialog '32de0b9d-c79fec9a-7fe39c4b@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:48:09] Retransmitting #9 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9fb5c8
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as5e3b968c
To: "Leon Zhong" <sip:4134@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.179
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] WARNING[9661]: chan_sip.c:3618 retrans_pkt: Retransmission timeout reached on transmission 80914c21-4966b340-c96dfc17@172.30.0.180 for seqno 107 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 16626ms with no response
[2011-11-03 06:48:09] Retransmitting #2 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK752f0e1f
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as0135ac39
To: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #7 (no NAT) to 172.30.0.199:5060:
NOTIFY sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3b46b825
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as48fa5dfc
To: "4116" <sip:4116@customer.com>;tag=EEC06C12-96421E89
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: c50f2da8-a32effef-f7b8b3e@172.30.0.199
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4116@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #3 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c103a38
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.184:5060 --->
SUBSCRIBE sip:4127@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKead6f953B2938858
From: "4123" <sip:4123@customer.com>;tag=61CA9FD3-D8EB30D8
To: <sip:4127@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 16a0ba78-aedee661-2e9fbe7e@172.30.0.184
Contact: <sip:4123@172.30.0.184:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4123", realm="asterisk", nonce="72a2ca1d", uri="sip:4127@customer.com:5060", response="38cd2d15a15eebc30fca5875ccd3d199", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (16 headers 0 lines) ---
[2011-11-03 06:48:09] Creating new subscription
[2011-11-03 06:48:09] Sending to 172.30.0.184:5060 (no NAT)
[2011-11-03 06:48:09] list_route: hop: <sip:4123@172.30.0.184:5060>
[2011-11-03 06:48:09] Found peer '4123' for '4123' from 172.30.0.184:5060
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.184:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.184:5060;branch=z9hG4bKead6f953B2938858;received=172.30.0.184
From: "4123" <sip:4123@customer.com>;tag=61CA9FD3-D8EB30D8
To: <sip:4127@customer.com>;tag=as1090a257
Call-ID: 16a0ba78-aedee661-2e9fbe7e@172.30.0.184
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4bcc7562"
Content-Length: 0


<------------>
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog '16a0ba78-aedee661-2e9fbe7e@172.30.0.184' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09] Retransmitting #5 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK19aa9e2b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f7495ee
To: "4136" <sip:4136@customer.com>;tag=25EFCB8E-216CE9E7
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 17bbd438-be5aca11-f485beaa@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Really destroying SIP dialog '197fee59-47d4d9d8-89034e9f@172.30.0.181' Method: SUBSCRIBE
[2011-11-03 06:48:09] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK754b4acf
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.171:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKf76a8dc6903F12A9
From: "4106" <sip:4106@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 9987ba38-2c04b7a3-e8c3dcda@172.30.0.171
Contact: <sip:4106@172.30.0.171:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (15 headers 0 lines) ---
[2011-11-03 06:48:09] Creating new subscription
[2011-11-03 06:48:09] Sending to 172.30.0.171:5060 (no NAT)
[2011-11-03 06:48:09] list_route: hop: <sip:4106@172.30.0.171:5060>
[2011-11-03 06:48:09] Found peer '4106' for '4106' from 172.30.0.171:5060
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.171:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.171:5060;branch=z9hG4bKf76a8dc6903F12A9;received=172.30.0.171
From: "4106" <sip:4106@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>;tag=as718cf42c
Call-ID: 9987ba38-2c04b7a3-e8c3dcda@172.30.0.171
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1910473b"
Content-Length: 0


<------------>
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog '9987ba38-2c04b7a3-e8c3dcda@172.30.0.171' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09] Retransmitting #2 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK40f2adbb
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as26cc82d5
To: "4135" <sip:4135@customer.com>;tag=3A0EBCA5-24D5B71A
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: 4c393b60-e9b9a2a9-b618f20e@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #2 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ae4517c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a3eacbf
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 172b465e7557c12463787e9b1fe120b1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.195:5060 --->
SUBSCRIBE sip:4134@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc61d8d4c464C25C1
From: "Office 131" <sip:4113@customer.com>;tag=97F1C92-D1492FFF
To: <sip:4134@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: d896689a-25aa0ea7-39edb4f8@172.30.0.195
Contact: <sip:4113@172.30.0.195:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (15 headers 0 lines) ---
[2011-11-03 06:48:09] Creating new subscription
[2011-11-03 06:48:09] Sending to 172.30.0.195:5060 (no NAT)
[2011-11-03 06:48:09] list_route: hop: <sip:4113@172.30.0.195:5060>
[2011-11-03 06:48:09] Found peer '4113' for '4113' from 172.30.0.195:5060
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.195:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.195:5060;branch=z9hG4bKc61d8d4c464C25C1;received=172.30.0.195
From: "Office 131" <sip:4113@customer.com>;tag=97F1C92-D1492FFF
To: <sip:4134@customer.com>;tag=as1b9d6eaa
Call-ID: d896689a-25aa0ea7-39edb4f8@172.30.0.195
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="639c5418"
Content-Length: 0


<------------>
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog 'd896689a-25aa0ea7-39edb4f8@172.30.0.195' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK754b4acf
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.170:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK9fe738997EB2D1E8
From: "Office 105" <sip:4131@customer.com>;tag=B5BC0B23-C870CE5A
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 3475b013-9e85010a-5d40047d@172.30.0.170
Contact: <sip:4131@172.30.0.170:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (15 headers 0 lines) ---
[2011-11-03 06:48:09] Creating new subscription
[2011-11-03 06:48:09] Sending to 172.30.0.170:5060 (no NAT)
[2011-11-03 06:48:09] list_route: hop: <sip:4131@172.30.0.170:5060>
[2011-11-03 06:48:09] Found peer '4131' for '4131' from 172.30.0.170:5060
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.170:5060;branch=z9hG4bK9fe738997EB2D1E8;received=172.30.0.170
From: "Office 105" <sip:4131@customer.com>;tag=B5BC0B23-C870CE5A
To: <sip:4147@customer.com>;tag=as76595672
Call-ID: 3475b013-9e85010a-5d40047d@172.30.0.170
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01f4747e"
Content-Length: 0


<------------>
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog '3475b013-9e85010a-5d40047d@172.30.0.170' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09] Retransmitting #7 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK690a926b
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09] Retransmitting #5 (no NAT) to 172.30.0.180:5060:
NOTIFY sip:4120@172.30.0.180:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK186417c9
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as0e9574c5
To: "Office 106" <sip:4120@customer.com>;tag=DE8FE644-8FD862CD
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: fde2cac4-4612074d-b2293156@172.30.0.180
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4120@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.168:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK54cb957b518CC1A
From: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
To: <sip:4119@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
Contact: <sip:4125@172.30.0.168:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4125", realm="asterisk", nonce="53d0c268", uri="sip:4119@customer.com:5060", response="ce5f7c257979952426971fa4e5e5b719", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (16 headers 0 lines) ---
[2011-11-03 06:48:09] Creating new subscription
[2011-11-03 06:48:09] Sending to 172.30.0.168:5060 (no NAT)
[2011-11-03 06:48:09] Found peer '4125' for '4125' from 172.30.0.168:5060
[2011-11-03 06:48:09] Looking for 4119 in default-local (domain customer.com:5060)
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog '9ab1d38a-dba856ed-2992dc4@172.30.0.168' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.168:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.168:5060;branch=z9hG4bK54cb957b518CC1A;received=172.30.0.168
From: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
To: <sip:4119@customer.com>;tag=as0585287b
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4119@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:09] set_destination: Parsing <sip:4125@172.30.0.168:5060> for address/port to send to
[2011-11-03 06:48:09] set_destination: set destination to 172.30.0.168:5060
[2011-11-03 06:48:09] Reliably Transmitting (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06ea9050
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as0585287b
To: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:09]
<--- SIP read from UDP:172.30.0.166:5060 --->
SUBSCRIBE sip:4148@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKe26e67e2D08248D
From: "4137" <sip:4137@customer.com>;tag=64514EDD-574841B8
To: <sip:4148@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 8c28ad86-c4c5a271-4bc8f48c@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4137", realm="asterisk", nonce="6caaa015", uri="sip:4148@customer.com:5060", response="e2e06ca9c1723035d773c589daf192ae", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:09] --- (16 headers 0 lines) ---
[2011-11-03 06:48:09] Creating new subscription
[2011-11-03 06:48:09] Sending to 172.30.0.166:5060 (no NAT)
[2011-11-03 06:48:09] list_route: hop: <sip:4137@172.30.0.166:5060>
[2011-11-03 06:48:09] Found peer '4137' for '4137' from 172.30.0.166:5060
[2011-11-03 06:48:09]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKe26e67e2D08248D;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=64514EDD-574841B8
To: <sip:4148@customer.com>;tag=as07dfaa8a
Call-ID: 8c28ad86-c4c5a271-4bc8f48c@172.30.0.166
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c66929b"
Content-Length: 0


<------------>
[2011-11-03 06:48:09] Scheduling destruction of SIP dialog '8c28ad86-c4c5a271-4bc8f48c@172.30.0.166' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:09] Retransmitting #1 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcb002e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as491ccb3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6045d38a5732d0883b2f11175ee5bd8e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.187:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK2457e5ceCA108C21
From: "4168" <sip:4168@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: db6bcec2-2e8acd45-563892bc@172.30.0.187
Contact: <sip:4168@172.30.0.187:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4168", realm="asterisk", nonce="31eea1d8", uri="sip:4147@customer.com:5060", response="fa67f819f7502dde56842722a694f832", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (16 headers 0 lines) ---
[2011-11-03 06:48:10] Creating new subscription
[2011-11-03 06:48:10] Sending to 172.30.0.187:5060 (no NAT)
[2011-11-03 06:48:10] list_route: hop: <sip:4168@172.30.0.187:5060>
[2011-11-03 06:48:10] Found peer '4168' for '4168' from 172.30.0.187:5060
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.187:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.187:5060;branch=z9hG4bK2457e5ceCA108C21;received=172.30.0.187
From: "4168" <sip:4168@customer.com>;tag=E2F1CFBF-A6B8894E
To: <sip:4147@customer.com>;tag=as355b23d4
Call-ID: db6bcec2-2e8acd45-563892bc@172.30.0.187
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38b102b0"
Content-Length: 0


<------------>
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog 'db6bcec2-2e8acd45-563892bc@172.30.0.187' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
OPTIONS sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6d6e7517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fbefaa9
To: <sip:4135@172.30.0.177:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 786c9a7415b8cda21c9a8a240572459e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #1 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06ea9050
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as0585287b
To: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog 'f89e3598-d42ca483-7eea913a@172.30.0.197' Method: SUBSCRIBE
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.163:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK75dcdcb771194164
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
Contact: <sip:4138@172.30.0.163:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4138", realm="asterisk", nonce="2a6bd63c", uri="sip:4109@customer.com:5060", response="f2a0fbc883a729393d2a0c0410888e80", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (16 headers 0 lines) ---
[2011-11-03 06:48:10] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:10] Looking for 4109 in default-local (domain customer.com:5060)
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog 'ddf9dab8-ab9de2a1-75ab5826@172.30.0.163' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.163:5060;branch=z9hG4bK75dcdcb771194164;received=172.30.0.163
From: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
To: <sip:4109@customer.com>;tag=as3c290715
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4109@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:10] set_destination: Parsing <sip:4138@172.30.0.163:5060> for address/port to send to
[2011-11-03 06:48:10] set_destination: set destination to 172.30.0.163:5060
[2011-11-03 06:48:10] Reliably Transmitting (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086a9bc2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog '661f2035-7aecdd92-22db8323@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37f2dc98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4c398fc2
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ec7297323e9356605f6584e502685de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2aeba898
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #1 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3ccc2a26
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0fe734da
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7bf42ed227c7867a242679b1165a8445@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #1 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e78a517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c6afcaa
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28965d7e4bb4d12118d538437bc60dcb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #4 (no NAT) to 172.30.0.175:5060:
OPTIONS sip:4140@172.30.0.175:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0d6c55e8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as31687a6a
To: <sip:4140@172.30.0.175:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 5c286d7722b24420669a05493094337b@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK754b4acf
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog '5c286d7722b24420669a05493094337b@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:10] Really destroying SIP dialog '7182916f-eb9ad43e-b01090d1@172.30.0.187' Method: SUBSCRIBE
[2011-11-03 06:48:10] Retransmitting #5 (no NAT) to 172.30.0.178:5060:
NOTIFY sip:4173@172.30.0.178:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK59343336
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as05e66040
To: "4173" <sip:4173@customer.com>;tag=C396D0CD-AF04DC4C
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: e9eab26c-47a293a3-e7e49552@172.30.0.178
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4173@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.179:5060 --->
SUBSCRIBE sip:4119@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK35e78a38246A83FF
From: "Leon Zhong" <sip:4134@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 781d0a2a-d7fdcde1-2843320@172.30.0.179
Contact: <sip:4134@172.30.0.179:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4134", realm="asterisk", nonce="0fa691ee", uri="sip:4119@customer.com:5060", response="7920aaf88c0517cfaf679fe073c8afbf", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (16 headers 0 lines) ---
[2011-11-03 06:48:10] Creating new subscription
[2011-11-03 06:48:10] Sending to 172.30.0.179:5060 (no NAT)
[2011-11-03 06:48:10] Found peer '4134' for '4134' from 172.30.0.179:5060
[2011-11-03 06:48:10] Looking for 4119 in default-local (domain customer.com:5060)
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog '781d0a2a-d7fdcde1-2843320@172.30.0.179' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.179:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.179:5060;branch=z9hG4bK35e78a38246A83FF;received=172.30.0.179
From: "Leon Zhong" <sip:4134@customer.com>;tag=30F788B8-C2E26E7F
To: <sip:4119@customer.com>;tag=as2e21d9a2
Call-ID: 781d0a2a-d7fdcde1-2843320@172.30.0.179
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4119@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:10] set_destination: Parsing <sip:4134@172.30.0.179:5060> for address/port to send to
[2011-11-03 06:48:10] set_destination: set destination to 172.30.0.179:5060
[2011-11-03 06:48:10] Reliably Transmitting (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK27d43db6
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as2e21d9a2
To: "Leon Zhong" <sip:4134@customer.com>;tag=30F788B8-C2E26E7F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 781d0a2a-d7fdcde1-2843320@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #2 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06ea9050
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as0585287b
To: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog '9708841f-46781d2c-65ddf445@172.30.0.165' Method: SUBSCRIBE
[2011-11-03 06:48:10] Retransmitting #8 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK003f9a11
Max-Forwards: 70
From: <sip:4125@customer.com>;tag=as6b13d463
To: "4136" <sip:4136@customer.com>;tag=63B39F95-B834A72E
Contact: <sip:4125@172.30.0.10:5060>
Call-ID: 5ed83a08-2c9c4be1-25515c7a@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4125">
<address uri="sip:4125@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #1 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086a9bc2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #4 (no NAT) to 172.30.0.182:5060:
OPTIONS sip:4128@172.30.0.182:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK12da038b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as11d913e0
To: <sip:4128@172.30.0.182:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2d74748b177a1f29295abd52538bb0eb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Really destroying SIP dialog '2d74748b177a1f29295abd52538bb0eb@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.173:5060 --->
SUBSCRIBE sip:4135@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK26980db9EC38C190
From: "4118" <sip:4118@customer.com>;tag=D30B1C9A-5D282441
To: <sip:4135@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: a7eae618-d47d6a1f-552d3ed6@172.30.0.173
Contact: <sip:4118@172.30.0.173:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4118", realm="asterisk", nonce="32b4602b", uri="sip:4135@customer.com:5060", response="5cecdd67a3a3c6d308670ac08fcc612c", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (16 headers 0 lines) ---
[2011-11-03 06:48:10] Creating new subscription
[2011-11-03 06:48:10] Sending to 172.30.0.173:5060 (no NAT)
[2011-11-03 06:48:10] Found peer '4118' for '4118' from 172.30.0.173:5060
[2011-11-03 06:48:10] Looking for 4135 in default-local (domain customer.com:5060)
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog 'a7eae618-d47d6a1f-552d3ed6@172.30.0.173' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.173:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.173:5060;branch=z9hG4bK26980db9EC38C190;received=172.30.0.173
From: "4118" <sip:4118@customer.com>;tag=D30B1C9A-5D282441
To: <sip:4135@customer.com>;tag=as2872693e
Call-ID: a7eae618-d47d6a1f-552d3ed6@172.30.0.173
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4135@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:10] set_destination: Parsing <sip:4118@172.30.0.173:5060> for address/port to send to
[2011-11-03 06:48:10] set_destination: set destination to 172.30.0.173:5060
[2011-11-03 06:48:10] Reliably Transmitting (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a894ef4
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as2872693e
To: "4118" <sip:4118@customer.com>;tag=D30B1C9A-5D282441
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: a7eae618-d47d6a1f-552d3ed6@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog '78031bfd-6cc68e94-cf2353cf@172.30.0.178' Method: SUBSCRIBE
[2011-11-03 06:48:10] Retransmitting #2 (no NAT) to 172.30.0.171:5060:
OPTIONS sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK20a3af7c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as498bf214
To: <sip:4106@172.30.0.171:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 60f0a912692004145d32b44a156d546d@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #5 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK78841311
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as771d3887
To: "Leon Zhong" <sip:4134@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.178:5060 --->
SUBSCRIBE sip:4109@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK9ab1d38aDBA856ED
From: "4173" <sip:4173@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.178
Contact: <sip:4173@172.30.0.178:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4173", realm="asterisk", nonce="363c75be", uri="sip:4109@customer.com:5060", response="9216e9ed44fa487dbaad537a79fe276c", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (16 headers 0 lines) ---
[2011-11-03 06:48:10] Creating new subscription
[2011-11-03 06:48:10] Sending to 172.30.0.178:5060 (no NAT)
[2011-11-03 06:48:10] list_route: hop: <sip:4173@172.30.0.178:5060>
[2011-11-03 06:48:10] Found peer '4173' for '4173' from 172.30.0.178:5060
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.178:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.178:5060;branch=z9hG4bK9ab1d38aDBA856ED;received=172.30.0.178
From: "4173" <sip:4173@customer.com>;tag=8EF217BE-B0414651
To: <sip:4109@customer.com>;tag=as57afdbdb
Call-ID: 78031bfd-6cc68e94-cf2353cf@172.30.0.178
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ebcf756"
Content-Length: 0


<------------>
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog '78031bfd-6cc68e94-cf2353cf@172.30.0.178' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10] Retransmitting #6 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK54133146
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as2406f454
To: "empty station" <sip:4119@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: e5747d-a10f1dc-810b9183@172.30.0.191
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #2 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086a9bc2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.172:5060:
NOTIFY sip:4136@172.30.0.172:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK09270bb1
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as5b55693b
To: "4136" <sip:4136@customer.com>;tag=193E678D-A59D8926
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: c5abf131-7d77bdca-ef559923@172.30.0.172
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4136@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog 'a54d81f3-8acc6c7a-dfc435d9@172.30.0.188' Method: SUBSCRIBE
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.199:5060:
OPTIONS sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7ae4517c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as5a3eacbf
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 172b465e7557c12463787e9b1fe120b1@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.168:5060:
NOTIFY sip:4125@172.30.0.168:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK06ea9050
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as0585287b
To: "4125" <sip:4125@customer.com>;tag=855409A8-ED043E93
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: 9ab1d38a-dba856ed-2992dc4@172.30.0.168
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4125@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.191:5060:
NOTIFY sip:4119@172.30.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK752f0e1f
Max-Forwards: 70
From: <sip:4115@customer.com>;tag=as0135ac39
To: "empty station" <sip:4119@customer.com>;tag=CC75A1B6-F8970A0D
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 6a185af0-a88708f7-7ef9e206@172.30.0.191
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4119@customer.com;method=SUBSCRIBE" />
<atom id="4115">
<address uri="sip:4115@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #1 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a894ef4
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as2872693e
To: "4118" <sip:4118@customer.com>;tag=D30B1C9A-5D282441
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: a7eae618-d47d6a1f-552d3ed6@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #5 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6dc953f5
Max-Forwards: 70
From: <sip:4108@customer.com>;tag=as0a17f176
To: "4135" <sip:4135@customer.com>;tag=902627F9-2BA4261E
Contact: <sip:4108@172.30.0.10:5060>
Call-ID: 569c1668-7cf1ab91-1b38956@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4108">
<address uri="sip:4108@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.177:5060:
NOTIFY sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK4074131d
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as440eeb94
To: "4135" <sip:4135@customer.com>;tag=80AD0937-7A655BE4
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 75ab5826-6d5f5fd7-bf46e004@172.30.0.177
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4135@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.197:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK40d1bcaf5A84C296
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>
CSeq: 2 SUBSCRIBE
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Authorization: Digest username="4115", realm="asterisk", nonce="5dbddde7", uri="sip:4140@customer.com:5060", response="e5c1f11d3e6eff428d91b7e39faa4e4a", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (16 headers 0 lines) ---
[2011-11-03 06:48:10] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:10] Looking for 4140 in default-local (domain customer.com:5060)
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog 'f76a8dc6-903f12a9-9987ba38@172.30.0.197' in 3610000 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK40d1bcaf5A84C296;received=172.30.0.197
From: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
To: <sip:4140@customer.com>;tag=as756b08e3
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:4140@172.30.0.10:5060>;expires=3600
Content-Length: 0


<------------>
[2011-11-03 06:48:10] set_destination: Parsing <sip:4115@172.30.0.197:5060> for address/port to send to
[2011-11-03 06:48:10] set_destination: set destination to 172.30.0.197:5060
[2011-11-03 06:48:10] Reliably Transmitting (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42ba305d
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #4 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2c103a38
Max-Forwards: 70
From: <sip:4138@customer.com>;tag=as1e18c6e3
To: "4118" <sip:4118@customer.com>;tag=A486DF20-9F0FA1A7
Contact: <sip:4138@172.30.0.10:5060>
Call-ID: 31caffa2-d80e65c9-ba5c76a0@172.30.0.173
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4138">
<address uri="sip:4138@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog '7452d01f-f76a8dc6-903f12a9@172.30.0.170' Method: SUBSCRIBE
[2011-11-03 06:48:10] Retransmitting #2 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5ac0e69a
Max-Forwards: 70
From: <sip:4119@customer.com>;tag=as64914f5d
To: "4124" <sip:4124@customer.com>;tag=30F788B8-C2E26E7F
Contact: <sip:4119@172.30.0.10:5060>
Call-ID: a76923a7-aea30b36-e90f478d@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4119">
<address uri="sip:4119@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.181:5060:
NOTIFY sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2f95cdd5
Max-Forwards: 70
From: <sip:4148@customer.com>;tag=as7f4777df
To: "Office 107" <sip:4141@customer.com>;tag=85F3A65D-EC4FBCBC
Contact: <sip:4148@172.30.0.10:5060>
Call-ID: ba6e4732-6f5ab8a9-c50f2da8@172.30.0.181
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4141@customer.com;method=SUBSCRIBE" />
<atom id="4148">
<address uri="sip:4148@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.166:5060 --->
SUBSCRIBE sip:4147@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKe80c7b38B6D2CA43
From: "4137" <sip:4137@customer.com>;tag=BCC232F3-2C13642E
To: <sip:4147@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 7191887e-7bf650e9-bf7eee84@172.30.0.166
Contact: <sip:4137@172.30.0.166:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (15 headers 0 lines) ---
[2011-11-03 06:48:10] Creating new subscription
[2011-11-03 06:48:10] Sending to 172.30.0.166:5060 (no NAT)
[2011-11-03 06:48:10] list_route: hop: <sip:4137@172.30.0.166:5060>
[2011-11-03 06:48:10] Found peer '4137' for '4137' from 172.30.0.166:5060
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.166:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.166:5060;branch=z9hG4bKe80c7b38B6D2CA43;received=172.30.0.166
From: "4137" <sip:4137@customer.com>;tag=BCC232F3-2C13642E
To: <sip:4147@customer.com>;tag=as1e096fcf
Call-ID: 7191887e-7bf650e9-bf7eee84@172.30.0.166
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d9f0621"
Content-Length: 0


<------------>
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog '7191887e-7bf650e9-bf7eee84@172.30.0.166' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10] Retransmitting #3 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK086a9bc2
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #4 (no NAT) to 172.30.0.179:5060:
NOTIFY sip:4134@172.30.0.179:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK07057c98
Max-Forwards: 70
From: <sip:4147@customer.com>;tag=as397e6250
To: "Leon Zhong" <sip:4134@customer.com>;tag=4DA30A13-7B1D0742
Contact: <sip:4147@172.30.0.10:5060>
Call-ID: c5f32c09-76ce4e08-4f93254f@172.30.0.179
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4134@customer.com;method=SUBSCRIBE" />
<atom id="4147">
<address uri="sip:4147@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #5 (no NAT) to 172.30.0.163:5060:
NOTIFY sip:4138@172.30.0.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK434b6e17
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as3c290715
To: "4138" <sip:4138@customer.com>;tag=8FD0A098-469A901
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: ddf9dab8-ab9de2a1-75ab5826@172.30.0.163
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4138@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10]
<--- SIP read from UDP:172.30.0.183:5060 --->
SUBSCRIBE sip:4138@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bKfcc6f7974ADFE948
From: "4147" <sip:4147@customer.com>;tag=1397F27C-53BA98E5
To: <sip:4138@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: a5c4cdd1-cce16f72-73aa9e6b@172.30.0.183
Contact: <sip:4147@172.30.0.183:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:10] --- (15 headers 0 lines) ---
[2011-11-03 06:48:10] Creating new subscription
[2011-11-03 06:48:10] Sending to 172.30.0.183:5060 (no NAT)
[2011-11-03 06:48:10] list_route: hop: <sip:4147@172.30.0.183:5060>
[2011-11-03 06:48:10] Found peer '4147' for '4147' from 172.30.0.183:5060
[2011-11-03 06:48:10]
<--- Transmitting (no NAT) to 172.30.0.183:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.183:5060;branch=z9hG4bKfcc6f7974ADFE948;received=172.30.0.183
From: "4147" <sip:4147@customer.com>;tag=1397F27C-53BA98E5
To: <sip:4138@customer.com>;tag=as077a00bb
Call-ID: a5c4cdd1-cce16f72-73aa9e6b@172.30.0.183
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4317008d"
Content-Length: 0


<------------>
[2011-11-03 06:48:10] Scheduling destruction of SIP dialog 'a5c4cdd1-cce16f72-73aa9e6b@172.30.0.183' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:10] Retransmitting #2 (no NAT) to 172.30.0.187:5060:
OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2dcb002e
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as491ccb3e
To: <sip:4168@172.30.0.187:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 6045d38a5732d0883b2f11175ee5bd8e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:10] Retransmitting #4 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK754b4acf
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Really destroying SIP dialog '61221967-8836730c-98ca2405@172.30.0.175' Method: SUBSCRIBE
[2011-11-03 06:48:10] Retransmitting #7 (no NAT) to 172.30.0.171:5060:
NOTIFY sip:4106@172.30.0.171:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK1b4c7e64
Max-Forwards: 70
From: <sip:4109@customer.com>;tag=as4d512a0c
To: "4106" <sip:4106@customer.com>;tag=F0E9528A-93312BFD
Contact: <sip:4109@172.30.0.10:5060>
Call-ID: 32135a3e-144ece01-4f95f3f0@172.30.0.171
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4106@customer.com;method=SUBSCRIBE" />
<atom id="4109">
<address uri="sip:4109@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #1 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42ba305d
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #2 (no NAT) to 172.30.0.173:5060:
NOTIFY sip:4118@172.30.0.173:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK0a894ef4
Max-Forwards: 70
From: <sip:4135@customer.com>;tag=as2872693e
To: "4118" <sip:4118@customer.com>;tag=D30B1C9A-5D282441
Contact: <sip:4135@172.30.0.10:5060>
Call-ID: a7eae618-d47d6a1f-552d3ed6@172.30.0.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
ontent-Type: application/xpidf+xml
Content-Length: 364

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4118@customer.com;method=SUBSCRIBE" />
<atom id="4135">
<address uri="sip:4135@customer.com;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #8 (no NAT) to 172.30.0.164:5060:
NOTIFY sip:4124@172.30.0.164:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK52f44f74
Max-Forwards: 70
From: <sip:4127@customer.com>;tag=as3579c775
To: "4124" <sip:4124@customer.com>;tag=54F74FCF-2A06041E
Contact: <sip:4127@172.30.0.10:5060>
Call-ID: 347c00d6-458f082d-8eca8c0c@172.30.0.164
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4124@customer.com;method=SUBSCRIBE" />
<atom id="4127">
<address uri="sip:4127@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:10] Retransmitting #4 (no NAT) to 172.30.0.177:5060:
OPTIONS sip:4135@172.30.0.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK6d6e7517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as7fbefaa9
To: <sip:4135@172.30.0.177:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 786c9a7415b8cda21c9a8a240572459e@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:11]
<--- SIP read from UDP:172.30.0.196:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK579b90835750F798
From: "4108" <sip:4108@customer.com>;tag=5D353B03-A043AC18
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2c5c6fc1-7900604e-ef5e99cf@172.30.0.196
Contact: <sip:4108@172.30.0.196:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:11] --- (15 headers 0 lines) ---
[2011-11-03 06:48:11] Creating new subscription
[2011-11-03 06:48:11] Sending to 172.30.0.196:5060 (no NAT)
[2011-11-03 06:48:11] list_route: hop: <sip:4108@172.30.0.196:5060>
[2011-11-03 06:48:11] Found peer '4108' for '4108' from 172.30.0.196:5060
[2011-11-03 06:48:11]
<--- Transmitting (no NAT) to 172.30.0.196:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.196:5060;branch=z9hG4bK579b90835750F798;received=172.30.0.196
From: "4108" <sip:4108@customer.com>;tag=5D353B03-A043AC18
To: <sip:4140@customer.com>;tag=as040a3cce
Call-ID: 2c5c6fc1-7900604e-ef5e99cf@172.30.0.196
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46bbf0b1"
Content-Length: 0


<------------>
[2011-11-03 06:48:11] Scheduling destruction of SIP dialog '2c5c6fc1-7900604e-ef5e99cf@172.30.0.196' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:11] NOTICE[9661]: chan_sip.c:25579 sip_poke_noanswer: Peer '4135' is now UNREACHABLE!  Last qualify: 0
[2011-11-03 06:48:11] Really destroying SIP dialog '786c9a7415b8cda21c9a8a240572459e@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:11] Really destroying SIP dialog '630510e5-2b854b8e-91a879ab@172.30.0.195' Method: SUBSCRIBE
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4145 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4140 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4124 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4120 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4122 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4168 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4138 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4171 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4149 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4123 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4134 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4136 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4173 (queued)
[2011-11-03 06:48:11]   == Extension Changed 4135[default-local] new state Unavailable for Notify User 4118 (queued)
[2011-11-03 06:48:11]
<--- SIP read from UDP:172.30.0.165:5060 --->
SUBSCRIBE sip:4140@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK986b9a201B387669
From: "Office 113" <sip:4171@customer.com>;tag=82D00A0D-5D5F20E2
To: <sip:4140@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 9526a6ce-74dcd5df-962ed7ec@172.30.0.165
Contact: <sip:4171@172.30.0.165:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:11] --- (15 headers 0 lines) ---
[2011-11-03 06:48:11] Creating new subscription
[2011-11-03 06:48:11] Sending to 172.30.0.165:5060 (no NAT)
[2011-11-03 06:48:11] list_route: hop: <sip:4171@172.30.0.165:5060>
[2011-11-03 06:48:11] Found peer '4171' for '4171' from 172.30.0.165:5060
[2011-11-03 06:48:11]
<--- Transmitting (no NAT) to 172.30.0.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.165:5060;branch=z9hG4bK986b9a201B387669;received=172.30.0.165
From: "Office 113" <sip:4171@customer.com>;tag=82D00A0D-5D5F20E2
To: <sip:4140@customer.com>;tag=as66d363d5
Call-ID: 9526a6ce-74dcd5df-962ed7ec@172.30.0.165
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2845c2f3"
Content-Length: 0


<------------>
[2011-11-03 06:48:11] Scheduling destruction of SIP dialog '9526a6ce-74dcd5df-962ed7ec@172.30.0.165' in 6400 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:11] Retransmitting #4 (no NAT) to 172.30.0.169:5060:
OPTIONS sip:4127@172.30.0.169:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK37f2dc98
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as4c398fc2
To: <sip:4127@172.30.0.169:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 2ec7297323e9356605f6584e502685de@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:11] Really destroying SIP dialog '2ec7297323e9356605f6584e502685de@172.30.0.10:5060' Method: OPTIONS
[2011-11-03 06:48:11] Retransmitting #2 (no NAT) to 172.30.0.186:5060:
OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3ccc2a26
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0fe734da
To: <sip:4169@172.30.0.186:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 7bf42ed227c7867a242679b1165a8445@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:11] Retransmitting #2 (no NAT) to 172.30.0.181:5060:
OPTIONS sip:4141@172.30.0.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2e78a517
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.30.0.10>;tag=as0c6afcaa
To: <sip:4141@172.30.0.181:5060>
Contact: <sip:asterisk@172.30.0.10:5060>
Call-ID: 28965d7e4bb4d12118d538437bc60dcb@172.30.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 03 Nov 2011 13:48:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2011-11-03 06:48:11] Retransmitting #2 (no NAT) to 172.30.0.197:5060:
NOTIFY sip:4115@172.30.0.197:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK42ba305d
Max-Forwards: 70
From: <sip:4140@customer.com>;tag=as756b08e3
To: "4115" <sip:4115@customer.com>;tag=1DEF41B1-179C3860
Contact: <sip:4140@172.30.0.10:5060>
Call-ID: f76a8dc6-903f12a9-9987ba38@172.30.0.197
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4115@customer.com;method=SUBSCRIBE" />
<atom id="4140">
<address uri="sip:4140@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:11] Retransmitting #6 (no NAT) to 172.30.0.192:5060:
NOTIFY sip:4148@172.30.0.192:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a9261d0
Max-Forwards: 70
From: <sip:4122@customer.com>;tag=as252ea80a
To: "4148" <sip:4148@customer.com>;tag=AC052B50-19938FDB
Contact: <sip:4122@172.30.0.10:5060>
Call-ID: cccbccfb-9f3dfab6-424f6aa1@172.30.0.192
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX (asterisk)
Subscription-State: active
Event: presence
Content-Type: application/xpidf+xml
Content-Length: 367

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:4148@customer.com;method=SUBSCRIBE" />
<atom id="4122">
<address uri="sip:4122@customer.com;user=ip" priority="0.800000">
<status status="closed" />
<msnsubstatus substatus="offline" />
</address>
</atom>
</presence>

---
[2011-11-03 06:48:11]
<--- SIP read from UDP:172.30.0.180:5060 --->
SUBSCRIBE sip:4135@customer.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK92a4d0a1B8DB293
From: "Office 106" <sip:4120@customer.com>;tag=D82EBBFA-79568983
To: <sip:4135@customer.com>
CSeq: 1 SUBSCRIBE
Call-ID: 2136859c-69f90625-8c09742e@172.30.0.180
Contact: <sip:4120@172.30.0.180:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Accept: application/xpidf+xml,text/xml+msrtc.pidf
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
[2011-11-03 06:48:11] --- (15 headers 0 lines) ---
[2011-11-03 06:48:11] Ignoring this SUBSCRIBE request
[2011-11-03 06:48:11] Found peer '4120' for '4120' from 172.30.0.180:5060
[2011-11-03 06:48:11]
<--- Transmitting (no NAT) to 172.30.0.180:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.180:5060;branch=z9hG4bK92a4d0a1B8DB293;received=172.30.0.180
From: "Office 106" <sip:4120@customer.com>;tag=D82EBBFA-79568983
To: <sip:4135@customer.com>;tag=as205f0fd6
Call-ID: 2136859c-69f90625-8c09742e@172.30.0.180
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="056dc74f"
Content-Length: 0


<------------>
[2011-11-03 06:48:11] Scheduling destruction of SIP dialog '2136859c-69f90625-8c09742e@172.30.0.180' in 9792 ms (Method: SUBSCRIBE)
[2011-11-03 06:48:11]