From: <sip:201@10.200.26.202;user=phone>;tag=as32cd7f7c
To: "Shawn Norton" <sip:710@10.200.26.202>;tag=f0b13b0adaad3b8c
Call-ID: bbf090d8cc415b76@10.200.26.110
CSeq: 117 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:710@10.200.26.110:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.103:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6bde863c;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as2c67a496
To: "Shelli Marvidakis" <sip:703@10.200.26.202>;tag=b57492549bc5dd38
Call-ID: 6347f968bfeba8a7@10.200.26.103
CSeq: 118 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:703@10.200.26.103:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
 Extension Changed 201 new state Idle for Notify User 727
Reliably Transmitting (no NAT) to 10.200.26.126:5060:
NOTIFY sip:726@10.200.26.126:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK146c1893;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as45fb996c
To: "Bob Teny" <sip:726@10.200.26.202>;tag=9e64b806dbe43e09
Contact: <sip:201@10.200.26.202>
Call-ID: 65e6ad5b684cc366@10.200.26.126
CSeq: 119 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="17" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 726
Reliably Transmitting (no NAT) to 10.200.26.125:5060:
NOTIFY sip:725@10.200.26.125:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK77f3b6c7;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as6b69a1b2
To: "Front Desk" <sip:725@10.200.26.202>;tag=d164f8f597fe9519
Contact: <sip:201@10.200.26.202>
Call-ID: 41c60b7a49eca8a9@10.200.26.125
CSeq: 119 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="17" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 725

<--- SIP read from 10.200.26.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3a882586;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as224c4582
To: "Foyer" <sip:730@10.200.26.202>;tag=15949c865d567ef2
Call-ID: 85d64dea473da1f5@10.200.26.130
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:730@10.200.26.130:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Reliably Transmitting (no NAT) to 10.200.26.124:5060:
NOTIFY sip:724@10.200.26.124:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b94ce0b;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as1f724fde
To: "Bobby Houston" <sip:724@10.200.26.202>;tag=d754fdb55c2e7b82
Contact: <sip:201@10.200.26.202>
Call-ID: 2eb6c41a6e9a2919@10.200.26.124
CSeq: 119 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="17" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 724

<--- SIP read from 10.200.26.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK32e7fe4d;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as1eae8923
To: "Will Jaimez" <sip:729@10.200.26.202>;tag=315438b5372e1682
Call-ID: 61b6080aa13a0cc5@10.200.26.129
CSeq: 120 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:729@10.200.26.129:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK65f8877b;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as5c667dd5
To: "Upstairs Kitchen" <sip:728@10.200.26.202>;tag=705477b5f62e7682
Call-ID: a4b66a0a233a8cc5@10.200.26.128
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:728@10.200.26.128:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Reliably Transmitting (no NAT) to 10.200.26.121:5060:
NOTIFY sip:721@10.200.26.121:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5e2a257a;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as0b9c86bf
To: "Danny D'Ambrosio" <sip:721@10.200.26.202>;tag=5c64b306f3e43409
Contact: <sip:201@10.200.26.202>
Call-ID: ced687fac3ade048@10.200.26.121
CSeq: 119 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205
asterisk-price*CLI>
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="17" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 721
Reliably Transmitting (no NAT) to 10.200.26.120:5060:
NOTIFY sip:720@10.200.26.120:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f4a5555;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as161e6555
To: "Darek Martinez" <sip:720@10.200.26.202>;tag=56643006f1e43309
Contact: <sip:201@10.200.26.202>
Call-ID: 0ad641fa8b9d4639@10.200.26.120
CSeq: 119 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="17" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 720
Reliably Transmitting (no NAT) to 10.200.26.118:5060:
NOTIFY sip:718@10.200.26.118:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ca4c893;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as0493f674
To: "Ed Hickman" <sip:718@10.200.26.202>;tag=75347c651cf55c19
Contact: <sip:201@10.200.26.202>
Call-ID: 63d60cea873d83f5@10.200.26.118
CSeq: 120 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="18" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 718
Reliably Transmitting (no NAT) to 10.200.26.109:5060:
NOTIFY sip:709@10.200.26.109:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1397a6a5;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as268a85bc
To: "Alan Young" <sip:709@10.200.26.202>;tag=3d81f832f91e1b1a
Contact: <sip:201@10.200.26.202>
Call-ID: 3e0174a522a9b67f@10.200.26.109
CSeq: 120 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="18" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 709

<--- SIP read from 10.200.26.127:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK50ece4f8;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as06a3fe0c
To: "Conference Room" <sip:727@10.200.26.202>;tag=7c64d40695e49609
Call-ID: a0e6695b254c8166@10.200.26.127
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:727@10.200.26.127:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.126:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK146c1893;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as45fb996c
To: "Bob Teny" <sip:726@10.200.26.202>;tag=9e64b806dbe43e09
Call-ID: 65e6ad5b684cc366@10.200.26.126
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:726@10.200.26.126:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Reliably Transmitting (no NAT) to 10.200.26.116:5060:
NOTIFY sip:716@10.200.26.116:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ec7b691;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as76ee1bdc
To: "Shop Kitchen" <sip:716@10.200.26.202>;tag=9934d275943676c2
Contact: <sip:201@10.200.26.202>
Call-ID: e0e6685b024c4c56@10.200.26.116
CSeq: 120 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="18" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 716

<--- SIP read from 10.200.26.124:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b94ce0b;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as1f724fde
To: "Bobby Houston" <sip:724@10.200.26.202>;tag=d754fdb55c2e7b82
Call-ID: 2eb6c41a6e9a2919@10.200.26.124
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:724@10.200.26.124:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.121:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5e2a257a;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as0b9c86bf
To: "Danny D'Ambrosio" <sip:721@10.200.26.202>;tag=5c64b306f3e43409
Call-ID: ced687fac3ade048@10.200.26.121
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:721@10.200.26.121:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.120:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f4a5555;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as161e6555
To: "Darek Martinez" <sip:720@10.200.26.202>;tag=56643006f1e43309
Call-ID: 0ad641fa8b9d4639@10.200.26.120
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:720@10.200.26.120:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.118:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ca4c893;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as0493f674
To: "Ed Hickman" <sip:718@10.200.26.202>;tag=75347c651cf55c19
Call-ID: 63d60cea873d83f5@10.200.26.118
CSeq: 120 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:718@10.200.26.118:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1397a6a5;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as268a85bc
To: "Alan Young" <sip:709@10.200.26.202>;tag=3d81f832f91e1b1a
Call-ID: 3e0174a522a9b67f@10.200.26.109
CSeq: 120 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:709@10.200.26.109:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ec7b691;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as76ee1bdc
To: "Shop Kitchen" <sip:716@10.200.26.202>;tag=9934d275943676c2
Call-ID: e0e6685b024c4c56@10.200.26.116
CSeq: 120 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:716@10.200.26.116:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.125:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK77f3b6c7;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as6b69a1b2
To: "Front Desk" <sip:725@10.200.26.202>;tag=d164f8f597fe9519
Call-ID: 41c60b7a49eca8a9@10.200.26.125
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:725@10.200.26.125:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Reliably Transmitting (no NAT) to 10.200.26.112:5060:
NOTIFY sip:712@10.200.26.112:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5fe96901;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as51a75cea
To: "Lorrie Blake" <sip:712@10.200.26.202>;tag=3534fc655df55e19
Contact: <sip:201@10.200.26.202>
Call-ID: 85d66dea283d63f5@10.200.26.112
CSeq: 120 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="18" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 712
Reliably Transmitting (no NAT) to 10.200.26.117:5060:
NOTIFY sip:717@10.200.26.117:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2d7a4bd4;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as45a134ad
To: "Ken Williams" <sip:717@10.200.26.202>;tag=1f2476b8763eb6b2
Contact: <sip:201@10.200.26.202>
Call-ID: 8cc6c38a6d4dc746@10.200.26.117
CSeq: 120 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="18" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 717
Reliably Transmitting (no NAT) to 10.200.26.115:5060:
NOTIFY sip:715@10.200.26.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7ac5cbf2;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as7b9bd37c
To: "Jerry Wright" <sip:715@10.200.26.202>;tag=3424bb255b259a82
Contact: <sip:201@10.200.26.202>
Call-ID: e8b68f0ac83aa2d5@10.200.26.115
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 715
Reliably Transmitting (no NAT) to 10.200.26.113:5060:
NOTIFY sip:713@10.200.26.113:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK79dbbafd;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as75385821
To: "Front Shop" <sip:713@10.200.26.202>;tag=53b1de0a5fad1f8c
Contact: <sip:201@10.200.26.202>
Call-ID: 5df0f2d8ef411d76@10.200.26.113
CSeq: 119 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="17" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 713
Reliably Transmitting (no NAT) to 10.200.26.108:5060:
NOTIFY sip:708@10.200.26.108:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK132d5b00;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as24940309
To: "Jake Ori" <sip:708@10.200.26.202>;tag=3e64fa14b205f2d1
Contact: <sip:201@10.200.26.202>
Call-ID: 413717f7fcdc8637@10.200.26.108
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 708
Reliably Transmitting (no NAT) to 10.200.26.107:5060:
NOTIFY sip:707@10.200.26.107:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78f32aee;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as53f0a8c1
To: "Tom Akers" <sip:707@10.200.26.202>;tag=b76414141bf47a88
Contact: <sip:201@10.200.26.202>
Call-ID: 212777877e8cc8a7@10.200.26.107
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---

<--- SIP read from 10.200.26.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4385cb59;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as431b36eb
To: "Jamie Stofko" <sip:731@10.200.26.202>;tag=c3d421ee56983dc3
Call-ID: fe373bf97a0e7a45@10.200.26.131
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:731@10.200.26.131:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
 Extension Changed 201 new state Idle for Notify User 707
Reliably Transmitting (no NAT) to 10.200.26.106:5060:
NOTIFY sip:706@10.200.26.106:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d8ac59c;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as792b1525
To: "Jenine Bentley" <sip:706@10.200.26.202>;tag=96fe78aeb32c7a87
Contact: <sip:201@10.200.26.202>
Call-ID: ff215e0951c9d32d@10.200.26.106
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 706
Reliably Transmitting (no NAT) to 10.200.26.105:5060:
NOTIFY sip:705@10.200.26.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1cbcb7d3;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as799e9b13
To: "Main Office Conf Room" <sip:705@10.200.26.202>;tag=d10e30137915bb02
Contact: <sip:201@10.200.26.202>
Call-ID: 0e4756783f5c6cb6@10.200.26.105
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 705
Reliably Transmitting (no NAT) to 10.200.26.104:5060:
NOTIFY sip:704@10.200.26.104:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1c49504a;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as2e9e1284
To: "Suzy Iorg" <sip:704@10.200.26.202>;tag=1c841b94b4a61618
Contact: <sip:201@10.200.26.202>
Call-ID: e95753e83eac6d46@10.200.26.104
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 704
Reliably Transmitting (no NAT) to 10.200.26.102:5060:
NOTIFY sip:702@10.200.26.102:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0461e104;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as77cad01b
To: "Jason Worley" <sip:702@10.200.26.202>;tag=9c843b9415a67718
Contact: <sip:201@10.200.26.202>
Call-ID: c067da49d45ba1d6@10.200.26.102
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---

<--- SIP read from 10.200.26.112:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5fe96901;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as51a75cea
To: "Lorrie Blake" <sip:712@10.200.26.202>;tag=3534fc655df55e19
Call-ID: 85d66dea283d63f5@10.200.26.112
CSeq: 120 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:712@10.200.26.112:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2d7a4bd4;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as45a134ad
To: "Ken Williams" <sip:717@10.200.26.202>;tag=1f2476b8763eb6b2
Call-ID: 8cc6c38a6d4dc746@10.200.26.117
CSeq: 120 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:717@10.200.26.117:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7ac5cbf2;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as7b9bd37c
To: "Jerry Wright" <sip:715@10.200.26.202>;tag=3424bb255b259a82
Call-ID: e8b68f0ac83aa2d5@10.200.26.115
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:715@10.200.26.115:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.113:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK79dbbafd;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as75385821
To: "Front Shop" <sip:713@10.200.26.202>;tag=53b1de0a5fad1f8c
Call-ID: 5df0f2d8ef411d76@10.200.26.113
CSeq: 119 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:713@10.200.26.113:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK132d5b00;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as24940309
To: "Jake Ori" <sip:708@10.200.26.202>;tag=3e64fa14b205f2d1
Call-ID: 413717f7fcdc8637@10.200.26.108
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:708@10.200.26.108:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78f32aee;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as53f0a8c1
To: "Tom Akers" <sip:707@10.200.26.202>;tag=b76414141bf47a88
Call-ID: 212777877e8cc8a7@10.200.26.107
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:707@10.200.26.107:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.106:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d8ac59c;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as792b1525
To: "Jenine Bentley" <sip:706@10.200.26.202>;tag=96fe78aeb32c7a87
Call-ID: ff215e0951c9d32d@10.200.26.106
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:706@10.200.26.106:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1cbcb7d3;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as799e9b13
To: "Main Office Conf Room" <sip:705@10.200.26.202>;tag=d10e30137915bb02
Call-ID: 0e4756783f5c6cb6@10.200.26.105
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:705@10.200.26.105:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.104:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1c49504a;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as2e9e1284
To: "Suzy Iorg" <sip:704@10.200.26.202>;tag=1c841b94b4a61618
Call-ID: e95753e83eac6d46@10.200.26.104
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:704@10.200.26.104:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
 Extension Changed 201 new state Idle for Notify User 702
Reliably Transmitting (no NAT) to 10.200.26.101:5060:
NOTIFY sip:701@10.200.26.101:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK66ff49d1;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as65b4d77f
To: "John Houston" <sip:701@10.200.26.202>;tag=b5847394fb96dc03
Contact: <sip:201@10.200.26.202>
Call-ID: 62577bda364c2366@10.200.26.101
CSeq: 121 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201@10.200.26.202">
<dialog id="201">
<state>terminated</state>
</dialog>
</dialog-info>

---
 Extension Changed 201 new state Idle for Notify User 701

<--- SIP read from 10.200.26.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0461e104;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as77cad01b
To: "Jason Worley" <sip:702@10.200.26.202>;tag=9c843b9415a67718
Call-ID: c067da49d45ba1d6@10.200.26.102
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:702@10.200.26.102:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 10.200.26.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK66ff49d1;rport
From: <sip:201@10.200.26.202;user=phone>;tag=as65b4d77f
To: "John Houston" <sip:701@10.200.26.202>;tag=b5847394fb96dc03
Call-ID: 62577bda364c2366@10.200.26.101
CSeq: 121 NOTIFY
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:701@10.200.26.101:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
asterisk-price*CLI>
<--- SIP read from 10.200.26.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK089ded08;rport
From: "SUBSCRIBER,WIRE" <sip:4358206441@10.200.26.202>;tag=as45fe9c3d
To: <sip:731@10.200.26.131:5060>;tag=0adacb34aab1f256
Call-ID: 07f08e2c55ea83764c3e62bd0ee58b7f@10.200.26.202
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:731@10.200.26.131:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 212

v=0
o=731 8000 8000 IN IP4 10.200.26.131
s=SIP Call
c=IN IP4 10.200.26.131
t=0 0
m=audio 5004 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.200.26.131:5004
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.200.26.131:5004
list_route: hop: <sip:731@10.200.26.131:5060>
set_destination: Parsing <sip:731@10.200.26.131:5060> for address/port to send to
set_destination: set destination to 10.200.26.131, port 5060
Transmitting (no NAT) to 10.200.26.131:5060:
ACK sip:731@10.200.26.131:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK10266eaa;rport
From: "SUBSCRIBER,WIRE" <sip:4358206441@10.200.26.202>;tag=as45fe9c3d
To: <sip:731@10.200.26.131:5060>;tag=0adacb34aab1f256
Contact: <sip:4358206441@10.200.26.202>
Call-ID: 07f08e2c55ea83764c3e62bd0ee58b7f@10.200.26.202
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- Call on SIP/731-082f2b38 left from hold
    -- SIP/731-082f2b38 answered Zap/2-1
asterisk-price*CLI>
<--- SIP read from 10.200.26.131:5060 --->
INVITE sip:4358206441@10.200.26.202 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bK006b2205c5615e0f
From: <sip:731@10.200.26.131:5060>;tag=0adacb34aab1f256
To: "SUBSCRIBER,WIRE" <sip:4358206441@10.200.26.202>;tag=as45fe9c3d
Contact: <sip:731@10.200.26.131:5060>
Supported: replaces, timer
Call-ID: 07f08e2c55ea83764c3e62bd0ee58b7f@10.200.26.202
CSeq: 20322 INVITE
User-Agent: Grandstream GXP2000 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 309

v=0
o=731 8000 8001 IN IP4 10.200.26.131
s=SIP Call
c=IN IP4 10.200.26.131
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3 101
a=sendonly
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
--- (13 headers 15 lines) ---
Sending to 10.200.26.131 : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 10.200.26.131:5004
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G723 for ID 4
Found description format G729 for ID 18
Found description format GSM for ID 3
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.200.26.131:5004
Audio is at 10.200.26.202 port 9022
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.200.26.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bK006b2205c5615e0f;received=10.200.26.131
From: <sip:731@10.200.26.131:5060>;tag=0adacb34aab1f256
To: "SUBSCRIBER,WIRE" <sip:4358206441@10.200.26.202>;tag=as45fe9c3d
Call-ID: 07f08e2c55ea83764c3e62bd0ee58b7f@10.200.26.202
CSeq: 20322 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:4358206441@10.200.26.202>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6669 6670 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 9022 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly

<------------>
    -- Started music on hold, class 'default', on Zap/2-1
asterisk-price*CLI>
<--- SIP read from 10.200.26.131:5060 --->
ACK sip:4358206441@10.200.26.202 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bK5a6b7d33908fa747
From: <sip:731@10.200.26.131:5060>;tag=0adacb34aab1f256
To: "SUBSCRIBER,WIRE" <sip:4358206441@10.200.26.202>;tag=as45fe9c3d
Contact: <sip:731@10.200.26.131:5060>
Call-ID: 07f08e2c55ea83764c3e62bd0ee58b7f@10.200.26.202
CSeq: 20322 ACK
User-Agent: Grandstream GXP2000 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
asterisk-price*CLI>
<--- SIP read from 10.200.26.131:5060 --->
REFER sip:4358206441@10.200.26.202 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bKd82d218631e50169
From: <sip:731@10.200.26.131:5060>;tag=0adacb34aab1f256
To: "SUBSCRIBER,WIRE" <sip:4358206441@10.200.26.202>;tag=as45fe9c3d
Contact: <sip:731@10.200.26.131:5060>
Supported: replaces
Refer-To: <sip:200@10.200.26.202>
Referred-By: <sip:731@10.200.26.202>
Call-ID: 07f08e2c55ea83764c3e62bd0ee58b7f@10.200.26.202
CSeq: 20323 REFER
User-Agent: Grandstream GXP2000 1.1.1.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Call 07f08e2c55ea83764c3e62bd0ee58b7f@10.200.26.202 got a SIP call transfer from caller: (REFER)!
SIP transfer to extension 200@from-internal by 731@10.200.26.202
    -- Starting simple switch on 'Zap/3-1'
    -- Executing [s@from-pstn:1] Answer("Zap/3-1", "") in new stack
    -- Executing [s@from-pstn:2] Goto("Zap/3-1", "790|1") in new stack
    -- Goto (from-pstn,790,1)
    -- Executing [790@from-pstn:1] Dial("Zap/3-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/707&SIP/708&SIP/709&SIP/712&SIP/715&SIP/716&SIP/717&SIP/718&SIP/720&SIP/721&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731|20|tr") in new stack
Audio is at 10.200.26.202 port 26182
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.101:5060:
INVITE sip:701@10.200.26.101:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK196a924c;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as66e7c7db
To: <sip:701@10.200.26.101:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 528c50f9317c4e34338bd31758a72408@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 26182 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 701
Audio is at 10.200.26.202 port 16246
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.103:5060:
INVITE sip:703@10.200.26.103:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07179c0b;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as29f4e41d
To: <sip:703@10.200.26.103:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 1e076bac6a45ce69792b59e85dbed549@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 16246 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 703
Audio is at 10.200.26.202 port 5392
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.104:5060:
INVITE sip:704@10.200.26.104:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK38b3d031;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as0275c0da
To: <sip:704@10.200.26.104:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 33d9adee798a112919f5e9976f9f67f6@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 5392 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 704I>
Audio is at 10.200.26.202 port 18054
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.105:5060:
INVITE sip:705@10.200.26.105:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK548e89bd;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as6c44c223
To: <sip:705@10.200.26.105:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 5c4f1f1231ab0a50059df3c3357e32dc@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 18054 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 705
Audio is at 10.200.26.202 port 16946
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.106:5060:
INVITE sip:706@10.200.26.106:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44c7fa8a;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as6d0bb35b
To: <sip:706@10.200.26.106:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 0dde83e6256b1d8d7c20009b233ea4b8@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 16946 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 706
Audio is at 10.200.26.202 port 9740
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.107:5060:
INVITE sip:707@10.200.26.107:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK58054c29;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as456f518f
To: <sip:707@10.200.26.107:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 7382a5c756a600fb0f22887a3f0589ea@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 9740 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 707I>
Audio is at 10.200.26.202 port 10134
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.108:5060:
INVITE sip:708@10.200.26.108:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK671d20da;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as72cc7431
To: <sip:708@10.200.26.108:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 2e6121cb0030008073fcc0b94a97e796@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 10134 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 708I>
Audio is at 10.200.26.202 port 14428
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.109:5060:
INVITE sip:709@10.200.26.109:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK71e8f693;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as270f02e7
To: <sip:709@10.200.26.109:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 58516e3f02f8be4f7be610be154c9a0b@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 14428 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 709I>
Audio is at 10.200.26.202 port 19648
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.112:5060:
INVITE sip:712@10.200.26.112:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16cc2329;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as67a2c562
To: <sip:712@10.200.26.112:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 006d94ba275ebfa955779ee304dd0dab@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 19648 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 712
Audio is at 10.200.26.202 port 27956
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.115:5060:
INVITE sip:715@10.200.26.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1b948896;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as5cc36a9e
To: <sip:715@10.200.26.115:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 74ec480d3e6438c93b67fc9d1112e479@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 27956 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 715I>
Audio is at 10.200.26.202 port 24240
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.116:5060:
INVITE sip:716@10.200.26.116:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21481bc5;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as1f925cdd
To: <sip:716@10.200.26.116:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 50b627ff4bcfd6454014de731218bbea@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 24240 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 716
Audio is at 10.200.26.202 port 13856
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.117:5060:
INVITE sip:717@10.200.26.117:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c9f682c;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as2c40afd3
To: <sip:717@10.200.26.117:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 5d80df3831f3a136205c4ccc7e646058@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 13856 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 717
Audio is at 10.200.26.202 port 9786
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.118:5060:
INVITE sip:718@10.200.26.118:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69e32778;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as11b8c466
To: <sip:718@10.200.26.118:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 2ca193dd223e82a21d79652a4e327a8e@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 9786 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 718
Audio is at 10.200.26.202 port 15934
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.120:5060:
INVITE sip:720@10.200.26.120:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK71db2046;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as72c54ed1
To: <sip:720@10.200.26.120:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 5925311669323d27755f49d324cafb37@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 15934 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 720
Audio is at 10.200.26.202 port 21464
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.121:5060:
INVITE sip:721@10.200.26.121:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK48cfa531;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as7e37cacc
To: <sip:721@10.200.26.121:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 2ca692c5166196640f3f227d47223880@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 21464 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 721I>
Audio is at 10.200.26.202 port 12440
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.124:5060:
INVITE sip:724@10.200.26.124:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK23229d7e;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as76809c89
To: <sip:724@10.200.26.124:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 6b9cc422466e9641153d291a2ea3a389@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 12440 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 724
Audio is at 10.200.26.202 port 22316
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.125:5060:
INVITE sip:725@10.200.26.125:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2bcd7ccf;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as73de22f9
To: <sip:725@10.200.26.125:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 02986ab66f9b5bab58845d5e56b9f91c@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 22316 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 725
Audio is at 10.200.26.202 port 8360
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.126:5060:
INVITE sip:726@10.200.26.126:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7c53f28a;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as444521c8
To: <sip:726@10.200.26.126:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 476c822361fe25575207676f46edb732@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 8360 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 726I>
Audio is at 10.200.26.202 port 8788
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.127:5060:
INVITE sip:727@10.200.26.127:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5f60cfc9;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as28099b5c
To: <sip:727@10.200.26.127:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 20f422e3527c6e863f6cb6c63d6fe37f@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 8788 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 727
Audio is at 10.200.26.202 port 21762
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.128:5060:
INVITE sip:728@10.200.26.128:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK74ec3ca5;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as7a775a65
To: <sip:728@10.200.26.128:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 71bf147904c5af4657e128863d155db1@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 21762 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 728I>
Audio is at 10.200.26.202 port 27194
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.129:5060:
INVITE sip:729@10.200.26.129:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK60aa1591;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as09bf2c15
To: <sip:729@10.200.26.129:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 4beb35f74f3dd95f72ecc99415001a20@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 27194 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 729I>
Audio is at 10.200.26.202 port 14242
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.130:5060:
INVITE sip:730@10.200.26.130:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f3b8c8c;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as37d1d7bb
To: <sip:730@10.200.26.130:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 448dc6bb57b2ff0208519c730722b473@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 14242 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 730
Audio is at 10.200.26.202 port 16808
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.26.131:5060:
INVITE sip:731@10.200.26.131:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK01e4b65f;rport
From: "INTERMOUNTAIN F" <sip:8014669035@10.200.26.202>;tag=as26260186
To: <sip:731@10.200.26.131:5060>
Contact: <sip:8014669035@10.200.26.202>
Call-ID: 730b090f5d6c12a93d2c661641b7b4bb@10.200.26.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 14 May 2007 19:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 6669 6669 IN IP4 10.200.26.202
s=session
c=IN IP4 10.200.26.202
t=0 0
m=audio 16808 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 731
    -- Nobody picked up in 20000 ms
    -- Executing [790@from-pstn:2] Goto("Zap/3-1", "791|1") in new stack
    -- Goto (from-pstn,791,1)
    -- Executing [791@from-pstn:1] Answer("Zap/3-1", "") in new stack
    -- Executing [791@from-pstn:2] SetMusicOnHold("Zap/3-1", "default") in new stack
    -- Executing [791@from-pstn:3] Set("Zap/3-1", "TIMEOUT(digit)=2") in new stack
    -- Digit timeout set to 2
    -- Executing [791@from-pstn:4] Set("Zap/3-1", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing [791@from-pstn:5] BackGround("Zap/3-1", "ie-greeting") in new stack
    -- <Zap/3-1> Playing 'ie-greeting' (language 'en')
  == CDR updated on Zap/3-1
    -- Executing [9@from-pstn:1] Directory("Zap/3-1", "default|from-internal|f") in new stack
  == Parsing '/etc/asterisk/voicemail.conf': Found
    -- <Zap/3-1> Playing 'dir-intro-fn' (language 'en')
  == Spawn extension (from-pstn, 9, 1) exited non-zero on 'Zap/3-1'
    -- Executing [h@from-pstn:1] Hangup("Zap/3-1", "") in new stack
  == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/3-1'
    -- Hungup 'Zap/3-1'
asterisk-price*CLI> [root@asterisk-price tmp]#