No audio on call forward after upgrade to 1.8

Asterisk Asterisk Support
1

Hi,

I am having the exact same issues as this post:

lists.digium.com/pipermail/aster … 53044.html

If someone sets call forwarding from their handset, there is no audio of ringtones that is played, and the call eventually drops instead of going to the voicemail or to an IVR

Never had this issue with 1.4. As soon as I upgraded to 1.8.15, we began to see this issue. Can anyone suggest anything? We are desperate for help.

The only temporary workaround is to “Answer” the call before it hits the original extension. But this will run us up a large bill with the carriers.

Thanks, and below is a debug


U 50.56.1.1:5060 -> 96.224.1.1:63419
INVITE sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK335a7106;rport.
Max-Forwards: 70.
From: "12124001111" <sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>.
Contact: <sip:12124001111@50.56.1.1:5060>.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 102 INVITE.
User-Agent: FPBX-2.10.0(1.8.15.0).
Date: Wed, 22 Aug 2012 15:31:39 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Diversion: <sip:23530202@50.56.1.1>;reason=unconditional.
Content-Type: application/sdp.
Content-Length: 284.
.
v=0.
o=root 2082520040 2082520040 IN IP4 50.56.1.1.
s=Asterisk PBX 1.8.15.0.
c=IN IP4 50.56.1.1.
t=0 0.
m=audio 19566 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=ptime:20.
a=sendrecv.


U 96.224.1.1:63419 -> 50.56.1.1:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK335a7106;rport=5060.
Contact: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
From: "12124001111"<sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 102 INVITE.
User-Agent: X-Lite release 5.0.0 stamp 67284.
Content-Length: 0.
.


U 96.224.1.1:63419 -> 50.56.1.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK335a7106;rport=5060.
Contact: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
From: "12124001111"<sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 102 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: X-Lite release 5.0.0 stamp 67284.
Content-Length: 211.
.
v=0.
o=- 12990123115229492 1 IN IP4 10.10.172.44.
s=CounterPath X-Lite 5.0.0.
c=IN IP4 10.10.172.44.
b=AS:1638.
t=0 0.
m=audio 5062 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 50.56.1.1:5060 -> 96.224.1.1:63419
ACK sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK5ef18a88;rport.
Max-Forwards: 70.
From: "12124001111" <sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
Contact: <sip:12124001111@50.56.1.1:5060>.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 102 ACK.
User-Agent: FPBX-2.10.0(1.8.15.0).
Content-Length: 0.
.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK04edade8;received=50.56.1.2;rport=5060.
From: "12124001111" <sip:12124001111@50.56.1.2>;tag=as17a96380.
To: <sip:13479185927@50.56.1.1>;tag=as49dcf6d2.
Call-ID: 75a585564bbb9c5c156ad83043262982@50.56.1.2.
CSeq: 102 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:13479185927@50.56.1.1:5060>.
Remote-Party-ID: "MBHA Test Extension" <sip:23530201@50.56.1.2>;party=called;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 1346694584 1346694584 IN IP4 50.56.1.1.
s=Asterisk PBX 1.8.15.0.
c=IN IP4 50.56.1.1.
t=0 0.
m=audio 13676 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 50.56.1.1:5060 -> 96.224.1.1:63419
BYE sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK31b98e2d;rport.
Max-Forwards: 70.
From: "12124001111" <sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 103 BYE.
User-Agent: FPBX-2.10.0(1.8.15.0).
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 96.224.1.1:63419
BYE sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK31b98e2d;rport.
Max-Forwards: 70.
From: "12124001111" <sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 103 BYE.
User-Agent: FPBX-2.10.0(1.8.15.0).
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 96.224.1.1:63419
BYE sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK31b98e2d;rport.
Max-Forwards: 70.
From: "12124001111" <sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 103 BYE.
User-Agent: FPBX-2.10.0(1.8.15.0).
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.


U 96.224.1.1:63419 -> 50.56.1.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK31b98e2d;rport=5060.
Contact: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
From: "12124001111"<sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 103 BYE.
User-Agent: X-Lite release 5.0.0 stamp 67284.
Content-Length: 0.
.


U 96.224.1.1:63419 -> 50.56.1.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK31b98e2d;rport=5060.
Contact: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
From: "12124001111"<sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 103 BYE.
User-Agent: X-Lite release 5.0.0 stamp 67284.
Content-Length: 0.
.


U 96.224.1.1:63419 -> 50.56.1.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK31b98e2d;rport=5060.
Contact: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>.
To: <sip:23530201@10.10.172.44:5060;rinstance=80cf54fd1459e68e>;tag=b1901fa2.
From: "12124001111"<sip:12124001111@50.56.1.1>;tag=as54d3e6e8.
Call-ID: 2caa20776a6a8a5560a9fc342ddf6e0d@50.56.1.1:5060.
CSeq: 103 BYE.
User-Agent: X-Lite release 5.0.0 stamp 67284.
Content-Length: 0.
.
2

The trace seems to be incomplete.

A failure on that update, for transfers is likely to indicate that the remote system doesn’t handle connected line update well. I seem to remember seeing that disabling calling party ID can work round that.

3
  • Are you referring to disabling Caller ID?
  • If so - when, and at which level? On the phone itself or through the asterisk config?
4

No.

Asterisk.

There are several ways of passing caller-ID. I believe that one of the extended methods is needed when doing connected line updates, and that, if you have a device with broken connected line update handling, setting sendrpid to no will disable the connected line update feature in Asterisk.

Connected line update was a much demanded feature.

You should search the archives for more details. As your trace seems incomplete, I can’t be sure that you have attempted a connected line update.

5

Thanks for your response. I tried setting sendprid and trustprid to no, on the trunk and all the extensions, but issue still occurring. Please see the below debug of a full call


U 50.56.1.2:5060 -> 50.56.1.1:5060
INVITE sip:12015551111@50.56.1.1 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK3c3e80b8;rport.
Max-Forwards: 70.
From: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>.
Contact: <sip:12124701111@50.56.1.2>.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.2.7.
Remote-Party-ID: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;privacy=off;screen=no.
Date: Thu, 23 Aug 2012 23:49:58 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 312.
.
v=0.
o=root 1595199209 1595199209 IN IP4 208.93.41.141.
s=Asterisk PBX 1.6.2.7.
c=IN IP4 208.93.41.141.
t=0 0.
m=audio 25692 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.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK3c3e80b8;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 102 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 47.23.104.218:31333
INVITE sip:38349205@10.10.2.51:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK533be7f0;rport.
Max-Forwards: 70.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as63f6a7b6.
To: <sip:38349205@10.10.2.51:5060>.
Contact: <sip:12124701111@50.56.1.1:5060>.
Call-ID: 6d7dc0721023bb9e328cc2ff748594fa@50.56.1.1:5060.
CSeq: 102 INVITE.
User-Agent: FPBX-2.10.0(1.8.15.0).
Date: Thu, 23 Aug 2012 23:49:57 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 282.
.
v=0.
o=root 657026096 657026096 IN IP4 50.56.1.1.
s=Asterisk PBX 1.8.15.0.
c=IN IP4 50.56.1.1.
t=0 0.
m=audio 19260 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=ptime:20.
a=sendrecv.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK3c3e80b8;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 102 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Length: 0.
.


U 47.23.104.218:31333 -> 50.56.1.1:5060
SIP/2.0 100 Trying.
To: <sip:38349205@10.10.2.51:5060>.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as63f6a7b6.
Call-ID: 6d7dc0721023bb9e328cc2ff748594fa@50.56.1.1:5060.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK533be7f0.
Server: Cisco/SPA504G-7.4.8a.
Content-Length: 0.
.


U 47.23.104.218:31333 -> 50.56.1.1:5060
SIP/2.0 302 Moved Temporarily.
To: <sip:38349205@10.10.2.51:5060>;tag=d940fc7ec34d3d11i0.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as63f6a7b6.
Call-ID: 6d7dc0721023bb9e328cc2ff748594fa@50.56.1.1:5060.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK533be7f0.
Contact: <sip:208@pbx01.cloudpoint.me>.
Diversion: <sip:38349205@pbx01.cloudpoint.me>;reason=unconditional.
Server: Cisco/SPA504G-7.4.8a.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 47.23.104.218:31333
ACK sip:38349205@10.10.2.51:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK533be7f0;rport.
Max-Forwards: 70.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as63f6a7b6.
To: <sip:38349205@10.10.2.51:5060>;tag=d940fc7ec34d3d11i0.
Contact: <sip:12124701111@50.56.1.1:5060>.
Call-ID: 6d7dc0721023bb9e328cc2ff748594fa@50.56.1.1:5060.
CSeq: 102 ACK.
User-Agent: FPBX-2.10.0(1.8.15.0).
Content-Length: 0.
.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 181 Call is being forwarded.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK3c3e80b8;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 102 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Diversion: <sip:38349205@50.56.1.1>;reason=unconditional.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 47.23.104.218:2458
INVITE sip:38349208@10.10.2.54:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK42e662ab;rport.
Max-Forwards: 70.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7350c0a4.
To: <sip:38349208@10.10.2.54:5060>.
Contact: <sip:12124701111@50.56.1.1:5060>.
Call-ID: 5bfe35c85ebdcd864fc7891f4dcfbf25@50.56.1.1:5060.
CSeq: 102 INVITE.
User-Agent: FPBX-2.10.0(1.8.15.0).
Date: Thu, 23 Aug 2012 23:49:58 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Diversion: <sip:38349205@50.56.1.1>;reason=unconditional.
Content-Type: application/sdp.
Content-Length: 282.
.
v=0.
o=root 390872339 390872339 IN IP4 50.56.1.1.
s=Asterisk PBX 1.8.15.0.
c=IN IP4 50.56.1.1.
t=0 0.
m=audio 15438 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=ptime:20.
a=sendrecv.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK3c3e80b8;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 102 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Length: 0.
.


U 47.23.104.218:2458 -> 50.56.1.1:5060
SIP/2.0 100 Trying.
To: <sip:38349208@10.10.2.54:5060>.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7350c0a4.
Call-ID: 5bfe35c85ebdcd864fc7891f4dcfbf25@50.56.1.1:5060.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK42e662ab.
Server: Cisco/SPA504G-7.4.8a.
Content-Length: 0.
.


U 47.23.104.218:2458 -> 50.56.1.1:5060
SIP/2.0 180 Ringing.
To: <sip:38349208@10.10.2.54:5060>;tag=e916aed37627ec2bi0.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7350c0a4.
Call-ID: 5bfe35c85ebdcd864fc7891f4dcfbf25@50.56.1.1:5060.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK42e662ab.
Contact: <sip:38349208@10.10.2.54:5060>.
Server: Cisco/SPA504G-7.4.8a.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 47.23.104.218:2458
CANCEL sip:38349208@10.10.2.54:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK42e662ab;rport.
Max-Forwards: 70.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7350c0a4.
To: <sip:38349208@10.10.2.54:5060>.
Call-ID: 5bfe35c85ebdcd864fc7891f4dcfbf25@50.56.1.1:5060.
CSeq: 102 CANCEL.
User-Agent: FPBX-2.10.0(1.8.15.0).
Content-Length: 0.
.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK3c3e80b8;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 102 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 1152331182 1152331182 IN IP4 50.56.1.1.
s=Asterisk PBX 1.8.15.0.
c=IN IP4 50.56.1.1.
t=0 0.
m=audio 10092 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 50.56.1.2:5060 -> 50.56.1.1:5060
ACK sip:12015551111@50.56.1.1:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK668f0397;rport.
Max-Forwards: 70.
From: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Contact: <sip:12124701111@50.56.1.2>.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.2.7.
Remote-Party-ID: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;privacy=off;screen=no.
Content-Length: 0.
.


U 50.56.1.2:5060 -> 50.56.1.1:5060
INVITE sip:12015551111@50.56.1.1:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK69f4e0db;rport.
Max-Forwards: 70.
From: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Contact: <sip:12124701111@50.56.1.2>.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX 1.6.2.7.
Remote-Party-ID: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;privacy=off;screen=no.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 288.
.
v=0.
o=root 1595199209 1595199210 IN IP4 208.93.41.141.
s=Asterisk PBX 1.6.2.7.
c=IN IP4 208.93.41.141.
t=0 0.
m=audio 25692 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK69f4e0db;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 103 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK69f4e0db;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 103 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 1152331182 1152331183 IN IP4 50.56.1.1.
s=Asterisk PBX 1.8.15.0.
c=IN IP4 50.56.1.1.
t=0 0.
m=audio 10092 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 50.56.1.2:5060 -> 50.56.1.1:5060
ACK sip:12015551111@50.56.1.1:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK31a34227;rport.
Max-Forwards: 70.
From: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Contact: <sip:12124701111@50.56.1.2>.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 103 ACK.
User-Agent: Asterisk PBX 1.6.2.7.
Remote-Party-ID: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;privacy=off;screen=no.
Content-Length: 0.
.


U 50.56.1.2:5060 -> 50.56.1.1:5060
INVITE sip:12015551111@50.56.1.1:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK4e6a80c1;rport.
Max-Forwards: 70.
From: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Contact: <sip:12124701111@50.56.1.2>.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 104 INVITE.
User-Agent: Asterisk PBX 1.6.2.7.
Remote-Party-ID: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;privacy=off;screen=no.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 288.
.
v=0.
o=root 1595199209 1595199211 IN IP4 208.93.41.141.
s=Asterisk PBX 1.6.2.7.
c=IN IP4 208.93.41.141.
t=0 0.
m=audio 25692 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK4e6a80c1;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 104 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK4e6a80c1;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 104 INVITE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Contact: <sip:12015551111@50.56.1.1:5060>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 1152331182 1152331184 IN IP4 50.56.1.1.
s=Asterisk PBX 1.8.15.0.
c=IN IP4 50.56.1.1.
t=0 0.
m=audio 10092 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 50.56.1.2:5060 -> 50.56.1.1:5060
ACK sip:12015551111@50.56.1.1:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK2ace03f8;rport.
Max-Forwards: 70.
From: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Contact: <sip:12124701111@50.56.1.2>.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 104 ACK.
User-Agent: Asterisk PBX 1.6.2.7.
Remote-Party-ID: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;privacy=off;screen=no.
Content-Length: 0.
.


U 50.56.1.2:5060 -> 50.56.1.1:5060
BYE sip:12015551111@50.56.1.1:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK6e6bb214;rport.
Max-Forwards: 70.
From: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 105 BYE.
User-Agent: Asterisk PBX 1.6.2.7.
Remote-Party-ID: "Cell Phone   NY" <sip:12124701111@50.56.1.2>;privacy=off;screen=no.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 50.56.1.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 50.56.1.2:5060;branch=z9hG4bK6e6bb214;received=50.56.1.2;rport=5060.
From: "Cell Phone NY" <sip:12124701111@50.56.1.2>;tag=as673d1448.
To: <sip:12015551111@50.56.1.1>;tag=as479dcdb7.
Call-ID: 78ef1f63146327d64c380a427f762f78@50.56.1.2.
CSeq: 105 BYE.
Server: FPBX-2.10.0(1.8.15.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces.
Content-Length: 0.
.


U 47.23.104.218:2458 -> 50.56.1.1:5060
SIP/2.0 487 Request Terminated.
To: <sip:38349208@10.10.2.54:5060>;tag=e916aed37627ec2bi0.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7350c0a4.
Call-ID: 5bfe35c85ebdcd864fc7891f4dcfbf25@50.56.1.1:5060.
CSeq: 102 INVITE.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK42e662ab.
Server: Cisco/SPA504G-7.4.8a.
Content-Length: 0.
.


U 50.56.1.1:5060 -> 47.23.104.218:2458
ACK sip:38349208@10.10.2.54:5060 SIP/2.0.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK42e662ab;rport.
Max-Forwards: 70.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7350c0a4.
To: <sip:38349208@10.10.2.54:5060>;tag=e916aed37627ec2bi0.
Contact: <sip:12124701111@50.56.1.1:5060>.
Call-ID: 5bfe35c85ebdcd864fc7891f4dcfbf25@50.56.1.1:5060.
CSeq: 102 ACK.
User-Agent: FPBX-2.10.0(1.8.15.0).
Content-Length: 0.
.


U 47.23.104.218:2458 -> 50.56.1.1:5060
SIP/2.0 200 OK.
To: <sip:38349208@10.10.2.54:5060>;tag=e916aed37627ec2bi0.
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7350c0a4.
Call-ID: 5bfe35c85ebdcd864fc7891f4dcfbf25@50.56.1.1:5060.
CSeq: 102 CANCEL.
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK42e662ab.
Server: Cisco/SPA504G-7.4.8a.
Content-Length: 0.
.
6

Could you please provide the trace from Asterisk, with debug level 5 and verbose 3 output as well. Analyzing these traces is always time consuming, but it is even more difficult when it is in an unfamiliar format, with few cues as to which is in and out, and none of the associated debugging and dialplan state data.

Also, please do the Google searches on the previous connected line ID problems, as I may have mis-remembered some of the details.

7

Thanks for your patience. I am posting 3 sets of debugs. Turning on debugging for the entire server at once will cause a huge mess.

Hope this will suffice

This is the trunk itself:


Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Contact: <sip:12124701111@50.56.1.3>
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7
Remote-Party-ID: "Cell Phone NY" <sip:12124701111@50.56.1.3>;privacy=off;screen=no
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:50.56.1.3:5060 --->
INVITE sip:12015551111@50.56.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK2c04b0b9;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Contact: <sip:12124701111@50.56.1.3>
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Remote-Party-ID: "Cell Phone NY" <sip:12124701111@50.56.1.3>;privacy=off;screen=no
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1360878165 1360878166 IN IP4 208.93.41.140
s=Asterisk PBX 1.6.2.7
c=IN IP4 208.93.41.140
t=0 0
m=audio 32324 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 50.56.1.3:5060 (NAT)
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=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 208.93.41.140:32324

<--- Transmitting (NAT) to 50.56.1.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK2c04b0b9;received=50.56.1.3;rport=5060
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 103 INVITE
Server: FPBX-2.10.0(1.8.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:12015551111@50.56.1.1:5060>
Content-Length: 0


<------------>
Audio is at 12548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 50.56.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK2c04b0b9;received=50.56.1.3;rport=5060
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 103 INVITE
Server: FPBX-2.10.0(1.8.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:12015551111@50.56.1.1:5060>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 889114124 889114125 IN IP4 50.56.1.1
s=Asterisk PBX 1.8.15.0
c=IN IP4 50.56.1.1
t=0 0
m=audio 12548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:50.56.1.3:5060 --->
ACK sip:12015551111@50.56.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK79686013;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Contact: <sip:12124701111@50.56.1.3>
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.7
Remote-Party-ID: "Cell Phone NY" <sip:12124701111@50.56.1.3>;privacy=off;screen=no
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:50.56.1.3:5060 --->
INVITE sip:12015551111@50.56.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK123d5c6e;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Contact: <sip:12124701111@50.56.1.3>
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Remote-Party-ID: "Cell Phone NY" <sip:12124701111@50.56.1.3>;privacy=off;screen=no
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1360878165 1360878167 IN IP4 208.93.41.140
s=Asterisk PBX 1.6.2.7
c=IN IP4 208.93.41.140
t=0 0
m=audio 32324 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 50.56.1.3:5060 (NAT)
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=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 208.93.41.140:32324

<--- Transmitting (NAT) to 50.56.1.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK123d5c6e;received=50.56.1.3;rport=5060
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 104 INVITE
Server: FPBX-2.10.0(1.8.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:12015551111@50.56.1.1:5060>
Content-Length: 0


<------------>
Audio is at 12548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 50.56.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK123d5c6e;received=50.56.1.3;rport=5060
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 104 INVITE
Server: FPBX-2.10.0(1.8.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:12015551111@50.56.1.1:5060>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 889114124 889114126 IN IP4 50.56.1.1
s=Asterisk PBX 1.8.15.0
c=IN IP4 50.56.1.1
t=0 0
m=audio 12548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:50.56.1.3:5060 --->
ACK sip:12015551111@50.56.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK7528f0b2;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Contact: <sip:12124701111@50.56.1.3>
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.6.2.7
Remote-Party-ID: "Cell Phone NY" <sip:12124701111@50.56.1.3>;privacy=off;screen=no
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:50.56.1.3:5060 --->
BYE sip:12015551111@50.56.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK489bd6f5;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.6.2.7
Remote-Party-ID: "Cell Phone NY" <sip:12124701111@50.56.1.3>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 50.56.1.3:5060 (NAT)
Scheduling destruction of SIP dialog '6d0ec30945fdd6943828fab57c9279d5@50.56.1.3' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 50.56.1.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 50.56.1.3:5060;branch=z9hG4bK489bd6f5;received=50.56.1.3;rport=5060
From: "Cell Phone NY" <sip:12124701111@50.56.1.3>;tag=as0f13fe97
To: <sip:12015551111@50.56.1.1>;tag=as1c69b7fa
Call-ID: 6d0ec30945fdd6943828fab57c9279d5@50.56.1.3
CSeq: 105 BYE
Server: FPBX-2.10.0(1.8.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
  == Spawn extension (ivr-3, s, 5) exited non-zero on 'SIP/CloudPointSIP01-00000ebd'
    -- Executing [h@ivr-3:1] Hangup("SIP/CloudPointSIP01-00000ebd", "") in new stack

This is the extension that is initiating the call forward from their phone:



From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7a04afb5
To: <sip:38349205@10.10.2.51:5060>
Contact: <sip:12124701111@50.56.1.1:5060>
Call-ID: 34eb8ce456b679eb7db09d933cdc7a87@50.56.1.1:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.15.0)
Date: Wed, 29 Aug 2012 22:18:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1252751885 1252751885 IN IP4 50.56.1.1
s=Asterisk PBX 1.8.15.0
c=IN IP4 50.56.1.1
t=0 0
m=audio 11402 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=ptime:20
a=sendrecv

---
    -- Called SIP/38349205

<--- SIP read from UDP:47.23.104.218:17282 --->
SIP/2.0 100 Trying
To: <sip:38349205@10.10.2.51:5060>
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7a04afb5
Call-ID: 34eb8ce456b679eb7db09d933cdc7a87@50.56.1.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK5047beb3
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:47.23.104.218:17282 --->
SIP/2.0 302 Moved Temporarily
To: <sip:38349205@10.10.2.51:5060>;tag=d940fc7ec34d3d11i0
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7a04afb5
Call-ID: 34eb8ce456b679eb7db09d933cdc7a87@50.56.1.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK5047beb3
Contact: <sip:208@pbx01.cloudpoint.me>
Diversion: <sip:38349205@pbx01.cloudpoint.me>;reason=unconditional
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Got SIP response 302 "Moved Temporarily" back from 47.23.104.218:17282
RDNIS for this call is 38349205 (reason unconditional)
Transmitting (NAT) to 47.23.104.218:17282:
ACK sip:38349205@10.10.2.51:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK5047beb3;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as7a04afb5
To: <sip:38349205@10.10.2.51:5060>;tag=d940fc7ec34d3d11i0
Contact: <sip:12124701111@50.56.1.1:5060>
Call-ID: 34eb8ce456b679eb7db09d933cdc7a87@50.56.1.1:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.0(1.8.15.0)
Content-Length: 0


---
    -- Now forwarding SIP/CloudPointSIP01-00000eba to 'Local/208@from-201-custom' (thanks to SIP/38349205-00000ebb)
    -- Executing [208@from-201-custom:1] Goto("Local/208@from-201-custom-6712;2", "from-clxfer-custom,208,1") in new stack
    -- Goto (from-clxfer-custom,208,1)
    -- Executing [208@from-clxfer-custom:1] Macro("Local/208@from-201-custom-6712;2", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("Local/208@from-201-custom-6712;2", "AMPUSER=12124701111") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("Local/208@from-201-custom-6712;2", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("Local/208@from-201-custom-6712;2", "1?Set(REALCALLERIDNUM=12124701111)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("Local/208@from-201-custom-6712;2", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("Local/208@from-201-custom-6712;2", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("Local/208@from-201-custom-6712;2", "1?report") in new stack
    -- Goto (macro-user-callerid,s,12)
    -- Executing [s@macro-user-callerid:12] GotoIf("Local/208@from-201-custom-6712;2", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:13] Set("Local/208@from-201-custom-6712;2", "__TTL=63") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("Local/208@from-201-custom-6712;2", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,25)
    -- Executing [s@macro-user-callerid:25] Set("Local/208@from-201-custom-6712;2", "CALLERID(number)=12124701111") in new stack
    -- Executing [s@macro-user-callerid:26] Set("Local/208@from-201-custom-6712;2", "CALLERID(name)=Cell Phone NY") in new stack
    -- Executing [s@macro-user-callerid:27] Set("Local/208@from-201-custom-6712;2", "CHANNEL(language)=en") in new stack
    -- Executing [208@from-clxfer-custom:2] Set("Local/208@from-201-custom-6712;2", "CALLERID(number)=12124701111") in new stack
    -- Executing [208@from-clxfer-custom:3] Goto("Local/208@from-201-custom-6712;2", "from-internal,38349208,1") in new stack
    -- Goto (from-internal,38349208,1)
    -- Executing [38349208@from-internal:1] Set("Local/208@from-201-custom-6712;2", "__RINGTIMER=20") in new stack
    -- Executing [38349208@from-internal:2] Macro("Local/208@from-201-custom-6712;2", "exten-vm,38349208,38349208,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("Local/208@from-201-custom-6712;2", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("Local/208@from-201-custom-6712;2", "AMPUSER=12124701111") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("Local/208@from-201-custom-6712;2", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("Local/208@from-201-custom-6712;2", "0?Set(REALCALLERIDNUM=12124701111)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("Local/208@from-201-custom-6712;2", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("Local/208@from-201-custom-6712;2", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("Local/208@from-201-custom-6712;2", "1?report") in new stack
    -- Goto (macro-user-callerid,s,12)
    -- Executing [s@macro-user-callerid:12] GotoIf("Local/208@from-201-custom-6712;2", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:13] Set("Local/208@from-201-custom-6712;2", "__TTL=62") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("Local/208@from-201-custom-6712;2", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,25)
    -- Executing [s@macro-user-callerid:25] Set("Local/208@from-201-custom-6712;2", "CALLERID(number)=12124701111") in new stack
    -- Executing [s@macro-user-callerid:26] Set("Local/208@from-201-custom-6712;2", "CALLERID(name)=Cell Phone NY") in new stack
    -- Executing [s@macro-user-callerid:27] Set("Local/208@from-201-custom-6712;2", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("Local/208@from-201-custom-6712;2", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("Local/208@from-201-custom-6712;2", "__EXTTOCALL=38349208") in new stack
    -- Executing [s@macro-exten-vm:4] Set("Local/208@from-201-custom-6712;2", "__PICKUPMARK=38349208") in new stack
    -- Executing [s@macro-exten-vm:5] Set("Local/208@from-201-custom-6712;2", "RT=20") in new stack
    -- Executing [s@macro-exten-vm:6] Gosub("Local/208@from-201-custom-6712;2", "sub-record-check,s,1(exten,38349208,)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("Local/208@from-201-custom-6712;2", "1?check") in new stack
    -- Goto (sub-record-check,s,6)
    -- Executing [s@sub-record-check:6] Set("Local/208@from-201-custom-6712;2", "__MON_FMT=gsm") in new stack
    -- Executing [s@sub-record-check:7] GotoIf("Local/208@from-201-custom-6712;2", "1?next") in new stack
    -- Goto (sub-record-check,s,10)
    -- Executing [s@sub-record-check:10] ExecIf("Local/208@from-201-custom-6712;2", "0?Return()") in new stack
    -- Executing [s@sub-record-check:11] GotoIf("Local/208@from-201-custom-6712;2", "1?exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] GotoIf("Local/208@from-201-custom-6712;2", "1?callee") in new stack
    -- Goto (sub-record-check,exten,8)
    -- Executing [exten@sub-record-check:8] GosubIf("Local/208@from-201-custom-6712;2", "0?record,1(exten,38349208,12124701111)") in new stack
    -- Executing [exten@sub-record-check:9] Return("Local/208@from-201-custom-6712;2", "") in new stack
    -- Executing [s@macro-exten-vm:7] GotoIf("Local/208@from-201-custom-6712;2", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,13)
    -- Executing [s@macro-exten-vm:13] GosubIf("Local/208@from-201-custom-6712;2", "0?clrheader,1()") in new stack
    -- Executing [s@macro-exten-vm:14] Macro("Local/208@from-201-custom-6712;2", "dial-one,20,r,38349208") in new stack
    -- Executing [s@macro-dial-one:1] Set("Local/208@from-201-custom-6712;2", "DEXTEN=38349208") in new stack
    -- Executing [s@macro-dial-one:2] Set("Local/208@from-201-custom-6712;2", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:3] GosubIf("Local/208@from-201-custom-6712;2", "0?screen,1()") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("Local/208@from-201-custom-6712;2", "0?cf,1()") in new stack
    -- Executing [s@macro-dial-one:5] GotoIf("Local/208@from-201-custom-6712;2", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,8)
    -- Executing [s@macro-dial-one:8] GotoIf("Local/208@from-201-custom-6712;2", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:9] GotoIf("Local/208@from-201-custom-6712;2", "0?continue") in new stack
    -- Executing [s@macro-dial-one:10] Set("Local/208@from-201-custom-6712;2", "EXTHASCW=ENABLED") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("Local/208@from-201-custom-6712;2", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,23)
    -- Executing [s@macro-dial-one:23] GotoIf("Local/208@from-201-custom-6712;2", "1?next3:continue") in new stack
    -- Goto (macro-dial-one,s,24)
    -- Executing [s@macro-dial-one:24] ExecIf("Local/208@from-201-custom-6712;2", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
    -- Executing [s@macro-dial-one:25] GotoIf("Local/208@from-201-custom-6712;2", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:26] GosubIf("Local/208@from-201-custom-6712;2", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("Local/208@from-201-custom-6712;2", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("Local/208@from-201-custom-6712;2", "DEVICES=38349208") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("Local/208@from-201-custom-6712;2", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("Local/208@from-201-custom-6712;2", "0?Set(DEVICES=8349208)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("Local/208@from-201-custom-6712;2", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("Local/208@from-201-custom-6712;2", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("Local/208@from-201-custom-6712;2", "THISDIAL=SIP/38349208") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("Local/208@from-201-custom-6712;2", "1?zap2dahdi,1()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("Local/208@from-201-custom-6712;2", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("Local/208@from-201-custom-6712;2", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("Local/208@from-201-custom-6712;2", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("Local/208@from-201-custom-6712;2", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("Local/208@from-201-custom-6712;2", "THISPART2=SIP/38349208") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("Local/208@from-201-custom-6712;2", "0?Set(THISPART2=DAHDI/38349208)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("Local/208@from-201-custom-6712;2", "NEWDIAL=SIP/38349208&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("Local/208@from-201-custom-6712;2", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("Local/208@from-201-custom-6712;2", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("Local/208@from-201-custom-6712;2", "THISDIAL=SIP/38349208") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("Local/208@from-201-custom-6712;2", "") in new stack
    -- Executing [dstring@macro-dial-one:9] Set("Local/208@from-201-custom-6712;2", "DSTRING=SIP/38349208&") in new stack
    -- Executing [dstring@macro-dial-one:10] Set("Local/208@from-201-custom-6712;2", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:11] GotoIf("Local/208@from-201-custom-6712;2", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:12] Set("Local/208@from-201-custom-6712;2", "DSTRING=SIP/38349208") in new stack
    -- Executing [dstring@macro-dial-one:13] Return("Local/208@from-201-custom-6712;2", "") in new stack
    -- Executing [s@macro-dial-one:27] GotoIf("Local/208@from-201-custom-6712;2", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("Local/208@from-201-custom-6712;2", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:29] GosubIf("Local/208@from-201-custom-6712;2", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("Local/208@from-201-custom-6712;2", "DB(CALLTRACE/38349208)=12124701111") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("Local/208@from-201-custom-6712;2", "") in new stack
    -- Executing [s@macro-dial-one:30] Set("Local/208@from-201-custom-6712;2", "D_OPTIONS=rM(auto-blkvm)") in new stack
    -- Executing [s@macro-dial-one:31] ExecIf("Local/208@from-201-custom-6712;2", "0?SIPAddHeader(Alert-Info: )") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("Local/208@from-201-custom-6712;2", "0?SIPAddHeader()") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("Local/208@from-201-custom-6712;2", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial-one:34] GosubIf("Local/208@from-201-custom-6712;2", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:35] Set("Local/208@from-201-custom-6712;2", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:36] Set("Local/208@from-201-custom-6712;2", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:37] GotoIf("Local/208@from-201-custom-6712;2", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:38] GotoIf("Local/208@from-201-custom-6712;2", "1?godial") in new stack
    -- Goto (macro-dial-one,s,42)
    -- Executing [s@macro-dial-one:42] Dial("Local/208@from-201-custom-6712;2", "SIP/38349208,20,rM(auto-blkvm)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Really destroying SIP dialog '34eb8ce456b679eb7db09d933cdc7a87@50.56.1.1:5060' Method: INVITE
    -- Called SIP/38349208
    -- Local/208@from-201-custom-6712;1 is ringing
    -- SIP/38349208-00000ebc is ringing
    -- Local/208@from-201-custom-6712;1 is ringing

<--- SIP read from UDP:47.23.104.218:17282 --->
NOTIFY sip:pbx01.cloudpoint.me SIP/2.0
Via: SIP/2.0/UDP 10.10.2.51:5060;branch=z9hG4bK-baecb7b5
From: <sip:38349205@pbx01.cloudpoint.me>;tag=82363d9a9f358ea5o0
To: <sip:pbx01.cloudpoint.me>
Call-ID: 2348bfde-3e095131@10.10.2.51
CSeq: 42451 NOTIFY
Max-Forwards: 70
Contact: <sip:38349205@10.10.2.51:5060>
Event: keep-alive
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (NAT) to 47.23.104.218:17282 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.2.51:5060;branch=z9hG4bK-baecb7b5;received=47.23.104.218;rport=17282
From: <sip:38349205@pbx01.cloudpoint.me>;tag=82363d9a9f358ea5o0
To: <sip:pbx01.cloudpoint.me>;tag=as33d29632
Call-ID: 2348bfde-3e095131@10.10.2.51
CSeq: 42451 NOTIFY
Server: FPBX-2.10.0(1.8.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2348bfde-3e095131@10.10.2.51' in 32000 ms (Method: NOTIFY)
    -- Nobody picked up in 15000 ms
  == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'Local/208@from-201-custom-6712;2' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'Local/208@from-201-custom-6712;2' in macro 'exten-vm'
  == Spawn extension (from-internal, 38349208, 2) exited non-zero on 'Local/208@from-201-custom-6712;2'
    -- Executing [h@from-internal:1] Hangup("Local/208@from-201-custom-6712;2", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/208@from-201-custom-6712;2'
    -- Executing [s@macro-dial:8] Set("SIP/CloudPointSIP01-00000eba", "DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/CloudPointSIP01-00000eba", "0?NOANSWER,1") in new stack
    -- Executing [38349601@ext-group:12] Gosub("SIP/CloudPointSIP01-00000eba", "sub-record-cancel,s,1()") in new stack
    -- Executing [s@sub-record-cancel:1] ExecIf("SIP/CloudPointSIP01-00000eba", "1?Return()") in new stack
    -- Executing [38349601@ext-group:13] Set("SIP/CloudPointSIP01-00000eba", "RingGroupMethod=") in new stack
    -- Executing [38349601@ext-group:14] GotoIf("SIP/CloudPointSIP01-00000eba", "0?nodest") in new stack
    -- Executing [38349601@ext-group:15] Set("SIP/CloudPointSIP01-00000eba", "__NODEST=") in new stack
    -- Executing [38349601@ext-group:16] Macro("SIP/CloudPointSIP01-00000eba", "blkvm-clr,") in new stack
    -- Executing [s@macro-blkvm-clr:1] Set("SIP/CloudPointSIP01-00000eba", "SHARED(BLKVM,SIP/CloudPointSIP01-00000eba)=") in new stack
    -- Executing [s@macro-blkvm-clr:2] Set("SIP/CloudPointSIP01-00000eba", "GOSUB_RETVAL=") in new stack
    -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/CloudPointSIP01-00000eba", "") in new stack
    -- Executing [38349601@ext-group:17] Goto("SIP/CloudPointSIP01-00000eba", "ivr-3,s,1") in new stack
    -- Goto (ivr-3,s,1)
    -- Executing [s@ivr-3:1] Set("SIP/CloudPointSIP01-00000eba", "_IVR_CONTEXT_ivr-3=") in new stack
    -- Executing [s@ivr-3:2] Set("SIP/CloudPointSIP01-00000eba", "_IVR_CONTEXT=ivr-3") in new stack
    -- Executing [s@ivr-3:3] Set("SIP/CloudPointSIP01-00000eba", "__IVR_RETVM=") in new stack
    -- Executing [s@ivr-3:4] GotoIf("SIP/CloudPointSIP01-00000eba", "0?skip") in new stack
    -- Executing [s@ivr-3:5] Answer("SIP/CloudPointSIP01-00000eba", "") in new stack
  == Spawn extension (ivr-3, s, 5) exited non-zero on 'SIP/CloudPointSIP01-00000eba'
    -- Executing [h@ivr-3:1] Hangup("SIP/CloudPointSIP01-00000eba", "") in new stack
  == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/CloudPointSIP01-00000eba'
Really destroying SIP dialog 'c1b18ce-887946d9@10.10.2.51' Method: REGISTER
cloudpoint-vpbx*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@cloudpoint-vpbx ~]#

This is the extension that is receiving the forwarded calls:




    -- Executing [s@macro-dial-one:37] GotoIf("Local/208@from-201-custom-b372;2", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:38] GotoIf("Local/208@from-201-custom-b372;2", "1?godial") in new stack
    -- Goto (macro-dial-one,s,42)
    -- Executing [s@macro-dial-one:42] Dial("Local/208@from-201-custom-b372;2", "SIP/38349208,20,rM(auto-blkvm)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 15616
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 47.23.104.218:30849:
INVITE sip:38349208@10.10.2.54:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK0160a02f;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as3996014b
To: <sip:38349208@10.10.2.54:5060>
Contact: <sip:12124701111@50.56.1.1:5060>
Call-ID: 5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.15.0)
Date: Wed, 29 Aug 2012 22:17:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Diversion: <sip:38349205@50.56.1.1>;reason=unconditional
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1875824769 1875824769 IN IP4 50.56.1.1
s=Asterisk PBX 1.8.15.0
c=IN IP4 50.56.1.1
t=0 0
m=audio 15616 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=ptime:20
a=sendrecv

---
    -- Called SIP/38349208
    -- Local/208@from-201-custom-b372;1 is ringing

<--- SIP read from UDP:47.23.104.218:30849 --->
SIP/2.0 100 Trying
To: <sip:38349208@10.10.2.54:5060>
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as3996014b
Call-ID: 5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK0160a02f
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:47.23.104.218:30849 --->
SIP/2.0 180 Ringing
To: <sip:38349208@10.10.2.54:5060>;tag=8b57cb117c16d528i0
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as3996014b
Call-ID: 5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK0160a02f
Contact: <sip:38349208@10.10.2.54:5060>
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:38349208@10.10.2.54:5060>
    -- SIP/38349208-00000eb9 is ringing
    -- Local/208@from-201-custom-b372;1 is ringing

<--- SIP read from UDP:47.23.104.218:30849 --->
NOTIFY sip:pbx01.cloudpoint.me SIP/2.0
Via: SIP/2.0/UDP 10.10.2.54:5060;branch=z9hG4bK-b93546c4
From: <sip:38349208@pbx01.cloudpoint.me>;tag=f6f1326b2dae36efo0
To: <sip:pbx01.cloudpoint.me>
Call-ID: 17d3ed9c-3e0acd73@10.10.2.54
CSeq: 45704 NOTIFY
Max-Forwards: 70
Contact: <sip:38349208@10.10.2.54:5060>
Event: keep-alive
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (NAT) to 47.23.104.218:30849 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.2.54:5060;branch=z9hG4bK-b93546c4;received=47.23.104.218;rport=30849
From: <sip:38349208@pbx01.cloudpoint.me>;tag=f6f1326b2dae36efo0
To: <sip:pbx01.cloudpoint.me>;tag=as078b857c
Call-ID: 17d3ed9c-3e0acd73@10.10.2.54
CSeq: 45704 NOTIFY
Server: FPBX-2.10.0(1.8.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '17d3ed9c-3e0acd73@10.10.2.54' in 32000 ms (Method: NOTIFY)
    -- Nobody picked up in 15000 ms
Scheduling destruction of SIP dialog '5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 47.23.104.218:30849:
CANCEL sip:38349208@10.10.2.54:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK0160a02f;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as3996014b
To: <sip:38349208@10.10.2.54:5060>
Call-ID: 5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.10.0(1.8.15.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060' in 6400 ms (Method: INVITE)
  == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'Local/208@from-201-custom-b372;2' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'Local/208@from-201-custom-b372;2' in macro 'exten-vm'
  == Spawn extension (from-internal, 38349208, 2) exited non-zero on 'Local/208@from-201-custom-b372;2'
    -- Executing [h@from-internal:1] Hangup("Local/208@from-201-custom-b372;2", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/208@from-201-custom-b372;2'
    -- Executing [s@macro-dial:8] Set("SIP/CloudPointSIP01-00000eb7", "DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/CloudPointSIP01-00000eb7", "0?NOANSWER,1") in new stack
    -- Executing [38349601@ext-group:12] Gosub("SIP/CloudPointSIP01-00000eb7", "sub-record-cancel,s,1()") in new stack
    -- Executing [s@sub-record-cancel:1] ExecIf("SIP/CloudPointSIP01-00000eb7", "1?Return()") in new stack
    -- Executing [38349601@ext-group:13] Set("SIP/CloudPointSIP01-00000eb7", "RingGroupMethod=") in new stack
    -- Executing [38349601@ext-group:14] GotoIf("SIP/CloudPointSIP01-00000eb7", "0?nodest") in new stack
    -- Executing [38349601@ext-group:15] Set("SIP/CloudPointSIP01-00000eb7", "__NODEST=") in new stack
    -- Executing [38349601@ext-group:16] Macro("SIP/CloudPointSIP01-00000eb7", "blkvm-clr,") in new stack
    -- Executing [s@macro-blkvm-clr:1] Set("SIP/CloudPointSIP01-00000eb7", "SHARED(BLKVM,SIP/CloudPointSIP01-00000eb7)=") in new stack
    -- Executing [s@macro-blkvm-clr:2] Set("SIP/CloudPointSIP01-00000eb7", "GOSUB_RETVAL=") in new stack
    -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/CloudPointSIP01-00000eb7", "") in new stack
    -- Executing [38349601@ext-group:17] Goto("SIP/CloudPointSIP01-00000eb7", "ivr-3,s,1") in new stack
    -- Goto (ivr-3,s,1)
    -- Executing [s@ivr-3:1] Set("SIP/CloudPointSIP01-00000eb7", "_IVR_CONTEXT_ivr-3=") in new stack
    -- Executing [s@ivr-3:2] Set("SIP/CloudPointSIP01-00000eb7", "_IVR_CONTEXT=ivr-3") in new stack
    -- Executing [s@ivr-3:3] Set("SIP/CloudPointSIP01-00000eb7", "__IVR_RETVM=") in new stack
    -- Executing [s@ivr-3:4] GotoIf("SIP/CloudPointSIP01-00000eb7", "0?skip") in new stack
    -- Executing [s@ivr-3:5] Answer("SIP/CloudPointSIP01-00000eb7", "") in new stack
  == Spawn extension (ivr-3, s, 5) exited non-zero on 'SIP/CloudPointSIP01-00000eb7'
    -- Executing [h@ivr-3:1] Hangup("SIP/CloudPointSIP01-00000eb7", "") in new stack
  == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/CloudPointSIP01-00000eb7'

<--- SIP read from UDP:47.23.104.218:30849 --->
SIP/2.0 487 Request Terminated
To: <sip:38349208@10.10.2.54:5060>;tag=8b57cb117c16d528i0
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as3996014b
Call-ID: 5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK0160a02f
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 47.23.104.218:30849:
ACK sip:38349208@10.10.2.54:5060 SIP/2.0
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK0160a02f;rport
Max-Forwards: 70
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as3996014b
To: <sip:38349208@10.10.2.54:5060>;tag=8b57cb117c16d528i0
Contact: <sip:12124701111@50.56.1.1:5060>
Call-ID: 5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.0(1.8.15.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:47.23.104.218:30849 --->
SIP/2.0 200 OK
To: <sip:38349208@10.10.2.54:5060>;tag=8b57cb117c16d528i0
From: "Cell Phone NY" <sip:12124701111@50.56.1.1>;tag=as3996014b
Call-ID: 5a2d96d43b5ec95413a06fb76ed37496@50.56.1.1:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 50.56.1.1:5060;branch=z9hG4bK0160a02f
Server: Cisco/SPA504G-7.4.8a
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
cloudpoint-vpbx*CLI> sip set debug off
SIP Debugging Disabled
8

Do you have the G.729 codec installed? How many licenses do you have for it?

9

Yes. About 10 licenses. But Ive tried this exact same thing with g711 and it was the exact same issue. If necessary I can post a sip debug in g711