INVITE sip:7323680452@172.16.4.4:5060 SIP/2.0
Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bK5mk2kb000o0gub4nd640.1
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
To: <sip:7323680452@135.25.29.135;user=phone>
CSeq: 7806 INVITE
Contact: <sip:+17323681000@135.25.29.135:5060;transport=udp>
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
Max-Forwards: 68
Content-Type: application/sdp
ontent-Length: 270
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Accept-Language: en; q=0.0
Allow: INVITE, BYE, ACK, CANCEL, PRACK, INFO
Content-Disposition: session; handling=required

v=0
o=Sonus_UAC 10845 6588 IN IP4 135.25.29.70
s=SIP Media Capabilities
c=IN IP4 135.25.29.70
t=0 0
m=audio 16754 RTP/AVP 2 18 0 96
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv

<------------->
--- (14 headers 12 lines) ---
Sending to 135.25.29.135 : 5060 (no NAT)
Using INVITE request as basis request - SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
Found peer 'first_gateway'
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 96
Peer audio RTP is at port 135.25.29.70:16754
Found description format G726-32 for ID 2
Found description format G729 for ID 18
Found description format PCMU for ID 0
Found description format telephone-event for ID 96
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x904 (ulaw|g726|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 135.25.29.70:16754
Looking for 7323680452 in pri_from_als (domain 172.16.4.4)
list_route: hop: <sip:+17323681000@135.25.29.135:5060;transport=udp>
thorium*CLI>
<--- Transmitting (no NAT) to 135.25.29.135:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bK5mk2kb000o0gub4nd640.1;received=135.25.29.135
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
To: <sip:7323680452@135.25.29.135;user=phone>
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 7806 INVITE
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7323680452@172.16.4.4>
Content-Length: 0


<------------>
    -- Executing [7323680452@pri_from_als:1] Macro("SIP/172.16.4.4-095b0460", "stdexten|0452|SIP/garrish") in new stack
    -- Executing [s@macro-stdexten:1] LDAPget("SIP/172.16.4.4-095b0460", "CIDNAME=name/||") in new stack
    -- LDAPget: varname=CIDNAME, config-section=name, keys=||
  == Parsing '/etc/asterisk/ldap.conf': Found
    -- LDAPget: ldap://www.post.att.com/o=att,c=us?name?subtree?(|(telephoneNumber=+)(telephoneNumber=+1)(mobileTelephoneNumber=+1))
    -- LDAPget: bind to www.post.att.com anonymously
    -- LDAPget: Value not found in directory.
    -- Executing [s@macro-stdexten:2] GotoIf("SIP/172.16.4.4-095b0460", "1?4:3") in new stack
    -- Goto (macro-stdexten,s,4)
    -- Executing [s@macro-stdexten:4] Dial("SIP/172.16.4.4-095b0460", "SIP/garrish|20") in new stack
Audio is at 172.16.4.4 port 23338
Reliably Transmitting (no NAT) to 172.16.4.26:5060:
NOTIFY sip:7325550455@172.16.4.26:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK00eb40dd;rport
From: <sip:0452@172.16.4.4:5060>;tag=as6f5bc143
To: 7325550455 <sip:7325550455@172.16.4.4:5060>;tag=ef61b9880e
Contact: <sip:0452@172.16.4.4>
Call-ID: 83c075b17910f43e
CSeq: 103 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 225

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:0452@172.16.4.4:5060">
<dialog id="0452" direction="recipient">
<state>early</state>
</dialog>
</dialog-info>

---
Adding codec 0x100 (g729) to SDP
 Extension Changed 0452 new state Ringing for Notify User 7325550455
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 172.16.4.23:5060:
NOTIFY sip:vince@172.16.4.23:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK23ce5dff;rport
From: <sip:0452@172.16.4.4>;tag=as0e482e5d
To: "7323680453" <sip:vince@172.16.4.4>;tag=9FC2EAEF-C14CC454
Contact: <sip:0452@172.16.4.4>
Call-ID: 10fc48b-7386b065-d5a55372@172.16.4.23
CSeq: 935 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 346

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:vince@172.16.4.4;method=SUBSCRIBE" />
<atom id="0452">
<address uri="sip:0452@172.16.4.4;user=ip" priority="0.800000">
<status status="inuse" />
<msnsubstatus substatus="onthephone" />
</address>
</atom>
</presence>

---
 Extension Changed 0452 new state Ringing for Notify User vince
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.4.21:5060:
NOTIFY sip:utano@172.16.4.21:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK0345f803;rport
From: <sip:0452@172.16.4.4>;tag=as73f6ab09
To: "7323680451" <sip:utano@172.16.4.4>;tag=27BFD56E-A73E8551
Contact: <sip:0452@172.16.4.4>
Call-ID: fa127a32-7101d168-a68c89e3@172.16.4.21
CSeq: 935 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 346

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:utano@172.16.4.4;method=SUBSCRIBE" />
<atom id="0452">
<address uri="sip:0452@172.16.4.4;user=ip" priority="0.800000">
<status status="inuse" />
<msnsubstatus substatus="onthephone" />
</address>
</atom>
</presence>

---
 Extension Changed 0452 new state Ringing for Notify User utano
Reliably Transmitting (no NAT) to 172.16.4.22:5060:
INVITE sip:garrish@172.16.4.22:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2bfae347;rport
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
To: <sip:garrish@172.16.4.22:5060>
Contact: <sip:+17323681000@172.16.4.4>
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
CSeq: 102 INVITE
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;privacy=off;screen=no
Date: Mon, 01 Oct 2007 20:49:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 32169 32169 IN IP4 135.25.29.70
s=session
c=IN IP4 135.25.29.70
t=0 0
m=audio 16754 RTP/AVP 18 0 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called garrish
thorium*CLI>
<--- SIP read from 172.16.4.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK0345f803;rport
From: <sip:0452@172.16.4.4>;tag=as73f6ab09
To: "7323680451" <sip:utano@172.16.4.4>;tag=27BFD56E-A73E8551
CSeq: 935 NOTIFY
Call-ID: fa127a32-7101d168-a68c89e3@172.16.4.21
Contact: <sip:utano@172.16.4.21:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
thorium*CLI>
<--- SIP read from 172.16.4.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK23ce5dff;rport
From: <sip:0452@172.16.4.4>;tag=as0e482e5d
To: "7323680453" <sip:vince@172.16.4.4>;tag=9FC2EAEF-C14CC454
CSeq: 935 NOTIFY
Call-ID: 10fc48b-7386b065-d5a55372@172.16.4.23
Contact: <sip:vince@172.16.4.23:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2bfae347;rport
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
To: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
CSeq: 102 INVITE
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
thorium*CLI>
<--- SIP read from 172.16.4.26:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK00eb40dd;rport=5060;received=172.16.4.4
From: <sip:0452@172.16.4.4:5060>;tag=as6f5bc143
To: 7325550455 <sip:7325550455@172.16.4.4:5060>;tag=ef61b9880e
Call-ID: 83c075b17910f43e
CSeq: 103 NOTIFY
Contact: 7325550455 <sip:7325550455@172.16.4.26:5060;transport=udp>
Server: Aastra 57iCT/2.1.0.2145
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2bfae347;rport
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
To: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
CSeq: 102 INVITE
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Allow-Events: talk,hold,conference
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/garrish-095e0a68 is ringing
thorium*CLI>
<--- Transmitting (no NAT) to 135.25.29.135:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bK5mk2kb000o0gub4nd640.1;received=135.25.29.135
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
To: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 7806 INVITE
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7323680452@172.16.4.4>
Content-Length: 0


<------------>
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2bfae347;rport
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
To: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
CSeq: 102 INVITE
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Type: application/sdp
Content-Length: 198

v=0
o=- 1191271743 1191271743 IN IP4 172.16.4.22
s=Polycom IP Phone
c=IN IP4 172.16.4.22
t=0 0
m=audio 16390 RTP/AVP 18 96
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 18
Found RTP audio format 96
Peer audio RTP is at port 172.16.4.22:16390
Found description format G729 for ID 18
Found description format telephone-event for ID 96
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.4.22:16390
list_route: hop: <sip:garrish@172.16.4.22:5060>
set_destination: Parsing <sip:garrish@172.16.4.22:5060> for address/port to send to
set_destination: set destination to 172.16.4.22, port 5060
Reliably Transmitting (no NAT) to 172.16.4.26:5060:
NOTIFY sip:7325550455@172.16.4.26:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK01ba18b6;rport
From: <sip:0452@172.16.4.4:5060>;tag=as6f5bc143
To: 7325550455 <sip:7325550455@172.16.4.4:5060>;tag=ef61b9880e
Contact: <sip:0452@172.16.4.4>
Call-ID: 83c075b17910f43e
CSeq: 104 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 207

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="2" state="full" entity="sip:0452@172.16.4.4:5060">
<dialog id="0452">
<state>confirmed</state>
</dialog>
</dialog-info>

---
Transmitting (no NAT) to 172.16.4.22:5060:
ACK sip:garrish@172.16.4.22:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2e33e193;rport
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
To: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
Contact: <sip:+17323681000@172.16.4.4>
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
CSeq: 102 ACK
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;privacy=off;screen=no
Content-Length: 0


---
    -- SIP/garrish-095e0a68 answered SIP/172.16.4.4-095b0460
 Extension Changed 0452 new state InUse for Notify User 7325550455
Audio is at 172.16.4.4 port 19432
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.4.23:5060:
NOTIFY sip:vince@172.16.4.23:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK3b4559f5;rport
From: <sip:0452@172.16.4.4>;tag=as0e482e5d
To: "7323680453" <sip:vince@172.16.4.4>;tag=9FC2EAEF-C14CC454
Contact: <sip:0452@172.16.4.4>
Call-ID: 10fc48b-7386b065-d5a55372@172.16.4.23
CSeq: 936 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 346

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:vince@172.16.4.4;method=SUBSCRIBE" />
<atom id="0452">
<address uri="sip:0452@172.16.4.4;user=ip" priority="0.800000">
<status status="inuse" />
<msnsubstatus substatus="onthephone" />
</address>
</atom>
</presence>

---

<--- Reliably Transmitting (no NAT) to 135.25.29.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bK5mk2kb000o0gub4nd640.1;received=135.25.29.135
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
To: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 7806 INVITE
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7323680452@172.16.4.4>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 32169 32169 IN IP4 172.16.4.22
s=session
c=IN IP4 172.16.4.22
t=0 0
m=audio 16390 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
 Extension Changed 0452 new state InUse for Notify User vince
    -- Native bridging SIP/172.16.4.4-095b0460 and SIP/garrish-095e0a68
Reliably Transmitting (no NAT) to 172.16.4.21:5060:
NOTIFY sip:utano@172.16.4.21:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2689504f;rport
From: <sip:0452@172.16.4.4>;tag=as73f6ab09
To: "7323680451" <sip:utano@172.16.4.4>;tag=27BFD56E-A73E8551
Contact: <sip:0452@172.16.4.4>
Call-ID: fa127a32-7101d168-a68c89e3@172.16.4.21
CSeq: 936 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 346

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:utano@172.16.4.4;method=SUBSCRIBE" />
<atom id="0452">
<address uri="sip:0452@172.16.4.4;user=ip" priority="0.800000">
<status status="inuse" />
<msnsubstatus substatus="onthephone" />
</address>
</atom>
</presence>

---
 Extension Changed 0452 new state InUse for Notify User utano
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
ACK sip:7323680452@172.16.4.4 SIP/2.0
Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bK56r56l00e0khfcg1a481.1
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
To: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
Max-Forwards: 68
CSeq: 7806 ACK
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
thorium*CLI>
<--- SIP read from 172.16.4.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK3b4559f5;rport
From: <sip:0452@172.16.4.4>;tag=as0e482e5d
To: "7323680453" <sip:vince@172.16.4.4>;tag=9FC2EAEF-C14CC454
CSeq: 936 NOTIFY
Call-ID: 10fc48b-7386b065-d5a55372@172.16.4.23
Contact: <sip:vince@172.16.4.23:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
thorium*CLI>
<--- SIP read from 172.16.4.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2689504f;rport
From: <sip:0452@172.16.4.4>;tag=as73f6ab09
To: "7323680451" <sip:utano@172.16.4.4>;tag=27BFD56E-A73E8551
CSeq: 936 NOTIFY
Call-ID: fa127a32-7101d168-a68c89e3@172.16.4.21
Contact: <sip:utano@172.16.4.21:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
thorium*CLI>
<--- SIP read from 172.16.4.26:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK01ba18b6;rport=5060;received=172.16.4.4
From: <sip:0452@172.16.4.4:5060>;tag=as6f5bc143
To: 7325550455 <sip:7325550455@172.16.4.4:5060>;tag=ef61b9880e
Call-ID: 83c075b17910f43e
CSeq: 104 NOTIFY
Contact: 7325550455 <sip:7325550455@172.16.4.26:5060;transport=udp>
Server: Aastra 57iCT/2.1.0.2145
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Scheduling destruction of SIP dialog '47bd523469b103f8624a0f7961166802@172.16.4.4' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 172.16.4.26:5060:
NOTIFY sip:7325550455@172.16.4.26:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK66ccf470;rport
From: "asterisk" <sip:asterisk@172.16.4.4>;tag=as19314c01
To: <sip:7325550455@172.16.4.26:5060;transport=udp>
Contact: <sip:asterisk@172.16.4.4>
Call-ID: 47bd523469b103f8624a0f7961166802@172.16.4.4
CSeq: 102 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90

Messages-Waiting: no
Message-Account: sip:asterisk@172.16.4.4
Voice-Message: 0/0 (0/0)

---
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
INVITE sip:+17323681000@172.16.4.4 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK22800257C75E450C
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
CSeq: 1 INVITE
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 198

v=0
o=- 1191271743 1191271744 IN IP4 172.16.4.22
s=Polycom IP Phone
c=IN IP4 172.16.4.22
t=0 0
m=audio 16390 RTP/AVP 18 96
a=sendonly
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000

<------------->
--- (14 headers 9 lines) ---
Sending to 172.16.4.22 : 5060 (no NAT)
Found RTP audio format 18
Found RTP audio format 96
Peer audio RTP is at port 172.16.4.22:16390
Found description format G729 for ID 18
Found description format telephone-event for ID 96
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.4.22:16390
thorium*CLI>
<--- Transmitting (no NAT) to 172.16.4.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK22800257C75E450C;received=172.16.4.22
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
CSeq: 1 INVITE
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:+17323681000@172.16.4.4>
Content-Length: 0


<------------>
Audio is at 172.16.4.4 port 23338
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
thorium*CLI>
<--- Transmitting (no NAT) to 172.16.4.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK22800257C75E450C;received=172.16.4.22
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
CSeq: 1 INVITE
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:+17323681000@172.16.4.4>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 32169 32170 IN IP4 135.25.29.70
s=session
c=IN IP4 135.25.29.70
t=0 0
m=audio 16754 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly

<------------>
set_destination: Parsing <sip:+17323681000@135.25.29.135:5060;transport=udp> for address/port to send to
set_destination: set destination to 135.25.29.135, port 5060
Audio is at 172.16.4.4 port 19432
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 135.25.29.135:5060:
INVITE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK23e98508;rport
From: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Contact: <sip:7323680452@172.16.4.4>
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 102 INVITE
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 32169 32170 IN IP4 172.16.4.4
s=session
c=IN IP4 172.16.4.4
t=0 0
m=audio 19432 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Started music on hold, class 'sip', on SIP/172.16.4.4-095b0460
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK23e98508;rport=5060
From: <sip:7323680452@172.16.4.4;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@135.25.29.135;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 102 INVITE


<------------->
--- (6 headers 0 lines) ---
thorium*CLI>
<--- SIP read from 172.16.4.26:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK66ccf470;rport=5060;received=172.16.4.4
From: "asterisk" <sip:asterisk@172.16.4.4>;tag=as19314c01
To: <sip:7325550455@172.16.4.26:5060;transport=udp>;tag=522460780
Call-ID: 47bd523469b103f8624a0f7961166802@172.16.4.4
CSeq: 102 NOTIFY
Contact: <sip:172.16.4.26:5060>
Server: Aastra 57iCT/2.1.0.2145
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '47bd523469b103f8624a0f7961166802@172.16.4.4' Method: NOTIFY
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK23e98508;rport=5060
From: <sip:7323680452@172.16.4.4;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@135.25.29.135;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 102 INVITE
Contact: <sip:+17323681000@135.25.29.135:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 240
ccept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Accept-Language: en; q=0.0
Allow: INVITE, BYE, ACK, CANCEL, PRACK, INFO
Content-Disposition: session; handling=required

v=0
o=Sonus_UAC 10845 6589 IN IP4 135.25.29.70
s=SIP Media Capabilities
c=IN IP4 135.25.29.70
t=0 0
m=audio 16754 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv

<------------->
--- (13 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 96
Peer audio RTP is at port 135.25.29.70:16754
Found description format G729 for ID 18
Found description format telephone-event for ID 96
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 135.25.29.70:16754
list_route: hop: <sip:+17323681000@135.25.29.135:5060;transport=udp>
set_destination: Parsing <sip:+17323681000@135.25.29.135:5060;transport=udp> for address/port to send to
set_destination: set destination to 135.25.29.135, port 5060
Transmitting (no NAT) to 135.25.29.135:5060:
ACK sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK0b43ec8f;rport
From: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Contact: <sip:7323680452@172.16.4.4>
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 102 ACK
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
ACK sip:+17323681000@172.16.4.4 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bKed80c82532E83072
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
CSeq: 1 ACK
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Max-Forwards: 70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 135.25.29.1:5060:
OPTIONS sip:135.25.29.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK6ff88c54;rport
From: "asterisk" <sip:asterisk@172.16.4.4>;tag=as027c9110
To: <sip:135.25.29.1>
Contact: <sip:asterisk@172.16.4.4>
Call-ID: 424ebabf2f66a2f96b8e4b045b8c95d9@172.16.4.4
CSeq: 102 OPTIONS
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Oct 2007 20:49:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #1 (no NAT) to 135.25.29.1:5060:
OPTIONS sip:135.25.29.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK6ff88c54;rport
From: "asterisk" <sip:asterisk@172.16.4.4>;tag=as027c9110
To: <sip:135.25.29.1>
Contact: <sip:asterisk@172.16.4.4>
Call-ID: 424ebabf2f66a2f96b8e4b045b8c95d9@172.16.4.4
CSeq: 102 OPTIONS
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Oct 2007 20:49:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
INVITE sip:+17323681000@172.16.4.4 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK8c7687033DEA5E28
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
CSeq: 2 INVITE
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 224

v=0
o=- 1191271743 1191271745 IN IP4 172.16.4.22
s=Polycom IP Phone
c=IN IP4 172.16.4.22
t=0 0
m=audio 16390 RTP/AVP 18 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (14 headers 10 lines) ---
Sending to 172.16.4.22 : 5060 (no NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.4.22:16390
Found description format G729 for ID 18
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.4.22:16390
thorium*CLI>
<--- Transmitting (no NAT) to 172.16.4.22:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK8c7687033DEA5E28;received=172.16.4.22
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
CSeq: 2 INVITE
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:+17323681000@172.16.4.4>
Content-Length: 0


<------------>
Audio is at 172.16.4.4 port 23338
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.16.4.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK8c7687033DEA5E28;received=172.16.4.22
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
CSeq: 2 INVITE
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:+17323681000@172.16.4.4>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 32169 32171 IN IP4 135.25.29.70
s=session
c=IN IP4 135.25.29.70
t=0 0
m=audio 16754 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
set_destination: Parsing <sip:+17323681000@135.25.29.135:5060;transport=udp> for address/port to send to
set_destination: set destination to 135.25.29.135, port 5060
Audio is at 172.16.4.4 port 19432
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 135.25.29.135:5060:
INVITE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK6ff840ba;rport
From: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Contact: <sip:7323680452@172.16.4.4>
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 103 INVITE
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 32169 32171 IN IP4 172.16.4.22
s=session
c=IN IP4 172.16.4.22
t=0 0
m=audio 16390 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Stopped music on hold on SIP/172.16.4.4-095b0460
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK6ff840ba;rport=5060
From: <sip:7323680452@172.16.4.4;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@135.25.29.135;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 103 INVITE


<------------->
--- (6 headers 0 lines) ---
set_destination: Parsing <sip:+17323681000@135.25.29.135:5060;transport=udp> for address/port to send to
set_destination: set destination to 135.25.29.135, port 5060
Audio is at 172.16.4.4 port 19432
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 135.25.29.135:5060:
INVITE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK035e52fb;rport
From: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Contact: <sip:7323680452@172.16.4.4>
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 104 INVITE
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 32169 32172 IN IP4 172.16.4.22
s=session
c=IN IP4 172.16.4.22
t=0 0
m=audio 16390 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK035e52fb;rport=5060
From: <sip:7323680452@172.16.4.4;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@135.25.29.135;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 104 INVITE


<------------->
--- (6 headers 0 lines) ---
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK035e52fb;rport=5060
From: <sip:7323680452@172.16.4.4;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@135.25.29.135;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 104 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
set_destination: Parsing <sip:+17323681000@135.25.29.135:5060;transport=udp> for address/port to send to
set_destination: set destination to 135.25.29.135, port 5060
Transmitting (no NAT) to 135.25.29.135:5060:
ACK sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK035e52fb;rport
From: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Contact: <sip:7323680452@172.16.4.4>
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 104 ACK
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '1d5d73704e5f636d70837ed909bdab04@172.16.4.4' in 6400 ms (Method: INVITE)
  == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'SIP/172.16.4.4-095b0460' in macro 'stdexten'
  == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'SIP/172.16.4.4-095b0460'
Scheduling destruction of SIP dialog 'SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32' in 576 ms (Method: ACK)
set_destination: Parsing <sip:+17323681000@135.25.29.135:5060;transport=udp> for address/port to send to
set_destination: set destination to 135.25.29.135, port 5060
Reliably Transmitting (no NAT) to 172.16.4.26:5060:
NOTIFY sip:7325550455@172.16.4.26:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK1c2eb8c0;rport
From: <sip:0452@172.16.4.4:5060>;tag=as6f5bc143
To: 7325550455 <sip:7325550455@172.16.4.4:5060>;tag=ef61b9880e
Contact: <sip:0452@172.16.4.4>
Call-ID: 83c075b17910f43e
CSeq: 105 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 208

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="3" state="full" entity="sip:0452@172.16.4.4:5060">
<dialog id="0452">
<state>terminated</state>
</dialog>
</dialog-info>

---
Reliably Transmitting (no NAT) to 135.25.29.135:5060:
BYE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK52fa0eaf;rport
From: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 105 BYE
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
 Extension Changed 0452 new state Idle for Notify User 7325550455
Reliably Transmitting (no NAT) to 172.16.4.23:5060:
NOTIFY sip:vince@172.16.4.23:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK1862bca6;rport
From: <sip:0452@172.16.4.4>;tag=as0e482e5d
To: "7323680453" <sip:vince@172.16.4.4>;tag=9FC2EAEF-C14CC454
Contact: <sip:0452@172.16.4.4>
Call-ID: 10fc48b-7386b065-d5a55372@172.16.4.23
CSeq: 937 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 341

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:vince@172.16.4.4;method=SUBSCRIBE" />
<atom id="0452">
<address uri="sip:0452@172.16.4.4;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
 Extension Changed 0452 new state Idle for Notify User vince
Reliably Transmitting (no NAT) to 172.16.4.21:5060:
NOTIFY sip:utano@172.16.4.21:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK1ddd1e60;rport
From: <sip:0452@172.16.4.4>;tag=as73f6ab09
To: "7323680451" <sip:utano@172.16.4.4>;tag=27BFD56E-A73E8551
Contact: <sip:0452@172.16.4.4>
Call-ID: fa127a32-7101d168-a68c89e3@172.16.4.21
CSeq: 937 NOTIFY
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Event: presence
Content-Type: application/xpidf+xml
Subscription-State: active
Content-Length: 341

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:utano@172.16.4.4;method=SUBSCRIBE" />
<atom id="0452">
<address uri="sip:0452@172.16.4.4;user=ip" priority="0.800000">
<status status="open" />
<msnsubstatus substatus="online" />
</address>
</atom>
</presence>

---
 Extension Changed 0452 new state Idle for Notify User utano
Retransmitting #1 (no NAT) to 135.25.29.135:5060:
BYE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK52fa0eaf;rport
From: <sip:7323680452@135.25.29.135;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 105 BYE
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK52fa0eaf;rport=5060
From: <sip:7323680452@172.16.4.4;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@135.25.29.135;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 105 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
thorium*CLI>
<--- SIP read from 135.25.29.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK52fa0eaf;rport=5060
From: <sip:7323680452@172.16.4.4;user=phone>;tag=as71b3299e
To: "OUT_OF_AREA" <sip:+17323681000@135.25.29.135;user=phone>;tag=SDtspfc01-1191271743-6755231191947266-11
Call-ID: SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32
CSeq: 105 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
thorium*CLI>
<--- SIP read from 172.16.4.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK1862bca6;rport
From: <sip:0452@172.16.4.4>;tag=as0e482e5d
To: "7323680453" <sip:vince@172.16.4.4>;tag=9FC2EAEF-C14CC454
CSeq: 937 NOTIFY
Call-ID: 10fc48b-7386b065-d5a55372@172.16.4.23
Contact: <sip:vince@172.16.4.23:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived

<--- SIP read from 172.16.4.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK1ddd1e60;rport
From: <sip:0452@172.16.4.4>;tag=as73f6ab09
To: "7323680451" <sip:utano@172.16.4.4>;tag=27BFD56E-A73E8551
CSeq: 937 NOTIFY
Call-ID: fa127a32-7101d168-a68c89e3@172.16.4.21
Contact: <sip:utano@172.16.4.21:5060>
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
thorium*CLI>
<--- SIP read from 172.16.4.26:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK1c2eb8c0;rport=5060;received=172.16.4.4
From: <sip:0452@172.16.4.4:5060>;tag=as6f5bc143
To: 7325550455 <sip:7325550455@172.16.4.4:5060>;tag=ef61b9880e
Call-ID: 83c075b17910f43e
CSeq: 105 NOTIFY
Contact: 7325550455 <sip:7325550455@172.16.4.26:5060;transport=udp>
Server: Aastra 57iCT/2.1.0.2145
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
ACK sip:+17323681000@172.16.4.4 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bKa28f1a7178562FAE
From: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
To: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
CSeq: 2 ACK
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Max-Forwards: 70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:garrish@172.16.4.22:5060> for address/port to send to
set_destination: set destination to 172.16.4.22, port 5060
Reliably Transmitting (no NAT) to 172.16.4.22:5060:
BYE sip:garrish@172.16.4.22:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK63abdf5b;rport
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
To: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
CSeq: 103 BYE
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;privacy=off;screen=no
Content-Length: 0


---
Scheduling destruction of SIP dialog '1d5d73704e5f636d70837ed909bdab04@172.16.4.4' in 6400 ms (Method: ACK)
thorium*CLI>
<--- SIP read from 172.16.4.22:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK63abdf5b;rport
From: "OUT_OF_AREA" <sip:+17323681000@172.16.4.4>;tag=as07ea660c
To: <sip:garrish@172.16.4.22:5060>;tag=85DC2E76-B0B37E19
CSeq: 103 BYE
Call-ID: 1d5d73704e5f636d70837ed909bdab04@172.16.4.4
Contact: <sip:garrish@172.16.4.22:5060>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1d5d73704e5f636d70837ed909bdab04@172.16.4.4' Method: ACK
Retransmitting #2 (no NAT) to 135.25.29.1:5060:
OPTIONS sip:135.25.29.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK6ff88c54;rport
From: "asterisk" <sip:asterisk@172.16.4.4>;tag=as027c9110
To: <sip:135.25.29.1>
Contact: <sip:asterisk@172.16.4.4>
Call-ID: 424ebabf2f66a2f96b8e4b045b8c95d9@172.16.4.4
CSeq: 102 OPTIONS
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Oct 2007 20:49:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog 'SDtspfc01-9753e4bb2f1d30173aff58c2a27f6d2b-7c6qh32' Method: ACK
Retransmitting #3 (no NAT) to 135.25.29.1:5060:
OPTIONS sip:135.25.29.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK6ff88c54;rport
From: "asterisk" <sip:asterisk@172.16.4.4>;tag=as027c9110
To: <sip:135.25.29.1>
Contact: <sip:asterisk@172.16.4.4>
Call-ID: 424ebabf2f66a2f96b8e4b045b8c95d9@172.16.4.4
CSeq: 102 OPTIONS
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Oct 2007 20:49:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #4 (no NAT) to 135.25.29.1:5060:
OPTIONS sip:135.25.29.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK6ff88c54;rport
From: "asterisk" <sip:asterisk@172.16.4.4>;tag=as027c9110
To: <sip:135.25.29.1>
Contact: <sip:asterisk@172.16.4.4>
Call-ID: 424ebabf2f66a2f96b8e4b045b8c95d9@172.16.4.4
CSeq: 102 OPTIONS
User-Agent: Thorium Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Oct 2007 20:49:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '424ebabf2f66a2f96b8e4b045b8c95d9@172.16.4.4' Method: OPTIONS
thorium*CLI>
<--- SIP read from 135.91.152.49:5060 --->
REGISTER sip:172.16.4.4:5060 SIP/2.0
Via: SIP/2.0/UDP 135.91.152.49:5060;branch=z9hG4bK72d936ac50AF5661
From: "Thorium_0453" <sip:polycom4@172.16.4.4>;tag=A835E5D2-339D33A5
To: <sip:polycom4@172.16.4.4>
CSeq: 1 REGISTER
Call-ID: f72c31ae-9110c610-a8d71d3@135.91.152.49
Contact: <sip:polycom4@135.91.152.49:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.1.2.0078
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 135.91.152.49 : 5060 (no NAT)
thorium*CLI>
<--- Transmitting (no NAT) to 135.91.152.49:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 135.91.152.49:5060;branch=z9hG4bK72d936ac50AF5661;received=135.91.152.49
From: "Thorium_0453" <sip:polycom4@172.16.4.4>;tag=A835E5D2-339D33A5
To: <sip:polycom4@172.16.4.4>
Call-ID: f72c31ae-9110c610-a8d71d3@135.91.152.49
CSeq: 1 REGISTER
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:polycom4@172.16.4.4>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 135.91.152.49:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 135.91.152.49:5060;branch=z9hG4bK72d936ac50AF5661;received=135.91.152.49
From: "Thorium_0453" <sip:polycom4@172.16.4.4>;tag=A835E5D2-339D33A5
To: <sip:polycom4@172.16.4.4>;tag=as6588d975
Call-ID: f72c31ae-9110c610-a8d71d3@135.91.152.49
CSeq: 1 REGISTER
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259d3147"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'f72c31ae-9110c610-a8d71d3@135.91.152.49' in 32000 ms (Method: REGISTER)
thorium*CLI>
<--- SIP read from 135.91.152.49:5060 --->
REGISTER sip:172.16.4.4:5060 SIP/2.0
Via: SIP/2.0/UDP 135.91.152.49:5060;branch=z9hG4bK744a7dd77958E2F4
From: "Thorium_0453" <sip:polycom4@172.16.4.4>;tag=A835E5D2-339D33A5
To: <sip:polycom4@172.16.4.4>
CSeq: 2 REGISTER
Call-ID: f72c31ae-9110c610-a8d71d3@135.91.152.49
Contact: <sip:polycom4@135.91.152.49:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.1.2.0078
Authorization: Digest username="polycom4", realm="asterisk", nonce="259d3147", uri="sip:172.16.4.4:5060", response="178cb68d277ccf3106ca393bbdb88c33", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 135.91.152.49 : 5060 (no NAT)

<--- Transmitting (no NAT) to 135.91.152.49:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 135.91.152.49:5060;branch=z9hG4bK744a7dd77958E2F4;received=135.91.152.49
From: "Thorium_0453" <sip:polycom4@172.16.4.4>;tag=A835E5D2-339D33A5
To: <sip:polycom4@172.16.4.4>
Call-ID: f72c31ae-9110c610-a8d71d3@135.91.152.49
CSeq: 2 REGISTER
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:polycom4@172.16.4.4>
Content-Length: 0


<------------>
thorium*CLI>
<--- Transmitting (no NAT) to 135.91.152.49:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 135.91.152.49:5060;branch=z9hG4bK744a7dd77958E2F4;received=135.91.152.49
From: "Thorium_0453" <sip:polycom4@172.16.4.4>;tag=A835E5D2-339D33A5
To: <sip:polycom4@172.16.4.4>;tag=as6588d975
Call-ID: f72c31ae-9110c610-a8d71d3@135.91.152.49
CSeq: 2 REGISTER
User-Agent: Thorium Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="259d3147"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'f72c31ae-9110c610-a8d71d3@135.91.152.49' in 32000 ms (Method: REGISTER)
thorium*CLI> exit
Executing last minute cleanups
Asterisk cleanly ending (0).
[root@thorium asterisk]#