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<body class='hmmessage'><div dir='ltr'>Matthew,<BR> <BR><u>SIP.conf</u><br>[100]<br>username=100<br>secret=password<br>type=friend<br>host=dynamic<br>nat=yes<br>canreinvite=no<br>insecure=port<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>allow=g729<br>context=asterisk<br>qualify=no<br> <br><u>dialplan</u><br>[asterisk]<br>exten => _X.,1,AGI(call.php)<br>exten => h,1,AGI(hangup.php)<br> <br>SIP Trace:<br>201.xxx.xxx.xxx = SIP Softphone which originates the call<br>xxx.xxx.xxx.xxx = Asterisk server<br>yyy.yyy.yyy.yyy = ITSP<br> <br><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>INVITE sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx><br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-181300058-1--d87543-;rport<br>Call-ID: 6601fe453f41d566<br>CSeq: 1 INVITE<br>Contact: <sip:100@201.xxx.xxx.xxx:5060><br>Max-Forwards: 70<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>Content-Type: application/sdp<br>User-Agent: eyeBeam release 3007n stamp 17816<br>Content-Length: 228<br> v=0<br>o=- 7847157 7847631 IN IP4 201.xxx.xxx.xxx<br>s=eyeBeam<br>c=IN IP4 201.xxx.xxx.xxx<br>t=0 0<br>m=audio 8614 RTP/AVP 0 101<br>a=alt:1 1 : 05A48429 0000007F 201.xxx.xxx.xxx 8614<br>a=fmtp:101 0-15<br>a=rtpmap:101 telephone-event/8000<br>a=sendrecv<BR><-------------><br>[Jun 3 13:11:27] --- (12 headers 10 lines) ---<br>[Jun 3 13:11:27] == Using SIP RTP CoS mark 5<br>[Jun 3 13:11:27] Sending to 201.xxx.xxx.xxx : 5060 (no NAT)<br>[Jun 3 13:11:27] Using INVITE request as basis request - 6601fe453f41d566<br>[Jun 3 13:11:27] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060<br>[Jun 3 13:11:27] <br><--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-181300058-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as1999a2fb<br>Call-ID: 6601fe453f41d566<br>CSeq: 1 INVITE<br>Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c9deee3"<br>Content-Length: 0<br> <------------><br>[Jun 3 13:11:27] Scheduling destruction of SIP dialog '6601fe453f41d566' in 32000 ms (Method: INVITE)<br>[Jun 3 13:11:27] <br><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>INVITE sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx><br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;rport<br>Call-ID: 6601fe453f41d566<br>CSeq: 2 INVITE<br>Contact: <sip:100@201.xxx.xxx.xxx:5060><br>Max-Forwards: 70<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>Content-Type: application/sdp<br>User-Agent: eyeBeam release 3007n stamp 17816<br>Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773456@xxx.xxx.xxx.xxx",response="e84d9090cfc8e60f94768583218ae0ad",algorithm=MD5<br>Content-Length: 228<br>v=0<br>o=- 7847157 7847631 IN IP4 201.xxx.xxx.xxx<br>s=eyeBeam<br>c=IN IP4 201.xxx.xxx.xxx<br>t=0 0<br>m=audio 8614 RTP/AVP 0 101<br>a=alt:1 1 : 05A48429 0000007F 201.xxx.xxx.xxx 8614<br>a=fmtp:101 0-15<br>a=rtpmap:101 telephone-event/8000<br>a=sendrecv<br><-------------><br> [Jun 3 13:11:27] --- (13 headers 10 lines) ---<br> [Jun 3 13:11:27] Sending to 201.xxx.xxx.xxx : 5060 (NAT)<br> [Jun 3 13:11:27] Using INVITE request as basis request - 6601fe453f41d566<br> [Jun 3 13:11:27] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060<br> [Jun 3 13:11:27] Found RTP audio format 0<br> [Jun 3 13:11:27] Found RTP audio format 101<br> [Jun 3 13:11:27] Found audio description format telephone-event for ID 101<br> [Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br> [Jun 3 13:11:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br> [Jun 3 13:11:27] Peer audio RTP is at port 201.xxx.xxx.xxx:8614<br> [Jun 3 13:11:27] Looking for 12127773456 in asterisk (domain xxx.xxx.xxx.xxx)<br> [Jun 3 13:11:27] list_route: hop: <sip:100@201.xxx.xxx.xxx:5060><br>[Jun 3 13:11:27] <br><--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>To: <sip:12127773456@xxx.xxx.xxx.xxx><br>Call-ID: 6601fe453f41d566<br>CSeq: 2 INVITE<br>Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Contact: <sip:12127773456@xxx.xxx.xxx.xxx><br>Content-Length: 0<br><------------><br>[Jun 3 13:11:27] -- Executing [12127773456@asterisk:1] AGI("SIP/100-000034d8", "call.php") in new stack<br>[Jun 3 13:11:27] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php<br>[Jun 3 13:11:28] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456)<br>[Jun 3 13:11:28] == Using SIP RTP CoS mark 5<br>[Jun 3 13:11:28] Audio is at xxx.xxx.xxx.xxx port 56248<br>[Jun 3 13:11:28] Adding codec 0x4 (ulaw) to SDP<br>[Jun 3 13:11:28] Adding non-codec 0x1 (telephone-event) to SDP<br>[Jun 3 13:11:28] Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060:<br>INVITE sip:12127773456@yyy.yyy.yyy.yyy SIP/2.0<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport<br>Max-Forwards: 70<br>From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>To: <sip:12127773456@yyy.yyy.yyy.yyy><br>Contact: <sip:100@xxx.xxx.xxx.xxx><br>Call-ID: <a href="mailto:07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx">07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx</a><br>CSeq: 102 INVITE<br>User-Agent: PBX<br>Date: Mon, 03 Jun 2013 13:11:28 IST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>Content-Length: 248<br> v=0<br>o=WP 1365830908 1365830908 IN IP4 xxx.xxx.xxx.xxx<br>s=PBX<br>c=IN IP4 xxx.xxx.xxx.xxx<br>t=0 0<br>m=audio 56248 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><strong>---<br>[Jun 3 13:11:28] -- Called yyy.yyy.yyy.yyy/12127773456</strong><br>[Jun 3 13:11:28] <br><--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060<br>From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>To: <sip:12127773456@yyy.yyy.yyy.yyy>;tag=gK029aaa8c<br>Call-ID: <a href="mailto:07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx">07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx</a><br>CSeq: 102 INVITE<br>Content-Length: 0<br><-------------><br>--- (7 headers 0 lines) ---<br><--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---><br>SIP/2.0 183 Session Progress<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060<br>From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>To: <sip:12127773456@yyy.yyy.yyy.yyy>;tag=gK029aaa8c<br>Call-ID: <a href="mailto:07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx">07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx</a><br>CSeq: 102 INVITE<br>Contact: <sip:12127773456@yyy.yyy.yyy.yyy:5060><br>Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS<br>Content-Length: 234<br>Content-Disposition: session; handling=required<br>Content-Type: application/sdp<br>v=0<br>o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy<br>s=SIP Media Capabilities<br>c=IN IP4 zzz.zzz.zzz.zzz<br>t=0 0<br>m=audio 21996 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv<br>a=maxptime:20<br><-------------><br>[Jun 3 13:11:31] --- (11 headers 11 lines) ---<br>[Jun 3 13:11:31] Found RTP audio format 0<br>[Jun 3 13:11:31] Found RTP audio format 101<br>[Jun 3 13:11:31] Found audio description format PCMU for ID 0<br>[Jun 3 13:11:31] Found audio description format telephone-event for ID 101<br>[Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>[Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>[Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996<br>[Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-000034d9 is making progress passing it to SIP/100-000034d8<br>[Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042<br>[Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP<br>[Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP<br>[Jun 3 13:11:31] <br><--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 183 Session Progress<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as654371b0<br>Call-ID: 6601fe453f41d566<br>CSeq: 2 INVITE<br>Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Contact: <sip:12127773456@xxx.xxx.xxx.xxx><br>Content-Type: application/sdp<br>Content-Length: 248<br>v=0<br>o=WP 1745765504 1745765504 IN IP4 xxx.xxx.xxx.xxx<br>s=PBX<br>c=IN IP4 xxx.xxx.xxx.xxx<br>t=0 0<br>m=audio 26042 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><------------><br>[0K[Jun 3 13:11:32] <br><--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060<br>From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>To: <sip:12127773456@yyy.yyy.yyy.yyy>;tag=gK029aaa8c<br>Call-ID: <a href="mailto:07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx">07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx</a><br>CSeq: 102 INVITE<br>Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed<br>Contact: <sip:12127773456@yyy.yyy.yyy.yyy:5060><br>Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS<br>Require: timer<br>Supported: timer,replaces<br>Session-Expires: 1800;refresher=uac<br>Content-Length: 234<br>Content-Disposition: session; handling=required<br>Content-Type: application/sdp<br>v=0<br>o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy<br>s=SIP Media Capabilities<br>c=IN IP4 zzz.zzz.zzz.zzz<br>t=0 0<br>m=audio 21996 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv<br>a=maxptime:20<br><-------------><br>[Jun 3 13:11:32] --- (15 headers 11 lines) ---<br>[Jun 3 13:11:32] list_route: hop: <sip:12127773456@yyy.yyy.yyy.yyy:5060><br>[Jun 3 13:11:32] set_destination: Parsing <sip:12127773456@yyy.yyy.yyy.yyy:5060> for address/port to send to<br>[Jun 3 13:11:32] set_destination: set destination to yyy.yyy.yyy.yyy, port 5060<br>[Jun 3 13:11:32] Transmitting (NAT) to yyy.yyy.yyy.yyy:5060:<br>ACK sip:12127773456@yyy.yyy.yyy.yyy:5060 SIP/2.0<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK570164a3;rport<br>Max-Forwards: 70<br>From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>To: <sip:12127773456@yyy.yyy.yyy.yyy>;tag=gK029aaa8c<br>Contact: <sip:100@xxx.xxx.xxx.xxx><br>Call-ID: <a href="mailto:07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx">07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx</a><br>CSeq: 102 ACK<br>User-Agent: PBX<br>Content-Length: 0<br> [Jun 3 13:11:32] -- SIP/yyy.yyy.yyy.yyy-000034d9 answered SIP/100-000034d8<br>[Jun 3 13:11:32] Audio is at xxx.xxx.xxx.xxx port 26042<br>[Jun 3 13:11:32] Adding codec 0x4 (ulaw) to SDP<br>[Jun 3 13:11:32] Adding non-codec 0x1 (telephone-event) to SDP<br>[Jun 3 13:11:32] <br><--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-254704504-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as654371b0<br>Call-ID: 6601fe453f41d566<br>CSeq: 2 INVITE<br>Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Contact: <sip:12127773456@xxx.xxx.xxx.xxx><br>Content-Type: application/sdp<br>Content-Length: 248<br>v=0<br>o=WP 1745765504 1745765505 IN IP4 xxx.xxx.xxx.xxx<br>s=PBX<br>c=IN IP4 xxx.xxx.xxx.xxx<br>t=0 0<br>m=audio 26042 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br>[Jun 3 13:11:32] -- Packet2Packet bridging SIP/100-000034d8 and SIP/yyy.yyy.yyy.yyy-000034d9<br>[Jun 3 13:11:32] <br><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>ACK sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as654371b0<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-558309054-1--d87543-;rport<br>Call-ID: 6601fe453f41d566<br>CSeq: 2 ACK<br>Contact: <sip:100@201.xxx.xxx.xxx:5060><br>Max-Forwards: 70<br>User-Agent: eyeBeam release 3007n stamp 17816<br>Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773456@xxx.xxx.xxx.xxx",response="e84d9090cfc8e60f94768583218ae0ad",algorithm=MD5<br>Content-Length: 0<br><-------------><br>[Jun 3 13:11:32] --- (11 headers 0 lines) ---<br>[Jun 3 13:11:32] Really destroying SIP dialog <a href="mailto:'2d60af2732e232cf3dec876f601c38d9@127.0.0.1'">'2d60af2732e232cf3dec876f601c38d9@127.0.0.1'</a> Method: REGISTER<br>[Jun 3 13:11:34] <br><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>BYE sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as654371b0<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-845749512-1--d87543-;rport<br>Call-ID: 6601fe453f41d566<br>CSeq: 3 BYE<br>Contact: <sip:100@201.xxx.xxx.xxx:5060><br>Max-Forwards: 70<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>User-Agent: eyeBeam release 3007n stamp 17816<br>Authorization: Digest username="100",realm="asterisk",nonce="0c9deee3",uri="sip:12127773456@xxx.xxx.xxx.xxx",response="ae82a7f181072735f72c23f2c63020e7",algorithm=MD5<br>Content-Length: 0<br>[Jun 3 13:11:35] --- (12 headers 0 lines) ---<br>[Jun 3 13:11:35] Sending to 201.xxx.xxx.xxx : 5060 (NAT)<br>[Jun 3 13:11:35] <br><--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-845749512-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=7c0c4b22<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as654371b0<br>Call-ID: 6601fe453f41d566<br>CSeq: 3 BYE<br>Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Length: 0<br> [Jun 3 13:11:35] -- Executing [h@asterisk:1] AGI("SIP/100-000034d8", "hangup.php") in new stack<br> [Jun 3 13:11:35] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php<br> [Jun 3 13:11:35] -- <SIP/100-000034d8>AGI Script hangup.php completed, returning 0<br> [Jun 3 13:11:35] Scheduling destruction of SIP dialog <a href="mailto:'07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx'">'07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx'</a> in 32000 ms (Method: INVITE)<br> [Jun 3 13:11:35] set_destination: Parsing <sip:12127773456@yyy.yyy.yyy.yyy:5060> for address/port to send to<br> [Jun 3 13:11:35] set_destination: set destination to yyy.yyy.yyy.yyy, port 5060<br> [Jun 3 13:11:35] Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060:<br>BYE sip:12127773456@yyy.yyy.yyy.yyy:5060 SIP/2.0<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport<br>Max-Forwards: 70<br>From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>To: <sip:12127773456@yyy.yyy.yyy.yyy>;tag=gK029aaa8c<br>Call-ID: <a href="mailto:07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx">07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx</a><br>CSeq: 103 BYE<br>User-Agent: PBX<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br> [Jun 3 13:11:35] -- <SIP/100-000034d8>AGI Script call.php completed, returning -1<br>[Jun 3 13:11:35] <br><--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK78ec518c;rport=5060<br>From: "100" <sip:100@xxx.xxx.xxx.xxx>;tag=as643c20b1<br>To: <sip:12127773456@yyy.yyy.yyy.yyy>;tag=gK029aaa8c<br>Call-ID: <a href="mailto:07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx">07714ae4593feb5c3e42b3a01cf4aa20@xxx.xxx.xxx.xxx</a><br>CSeq: 103 BYE<br>Content-Length: 0<BR>Regards,<br>Kamlesh<br><br> <BR><div>> Date: Fri, 31 May 2013 08:50:38 -0500<br>> From: mroth@imminc.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] G.729 codec in pass-thru mode<br>> <br>> Kamlesh Kumar wrote:<br>> > <br>> > Yes that's correct, when I use u-law call works fine.<br>> > <br>> > In case of g729, I enabled sip debug with 'sip set debug on' and captured all<br>> > the sip traces and got whatever I posted in last email. There was no other<br>> > call on the system when I captured sip trace. Please suggest further<br>> > proceedings. <br>> <br>> <br>> Kamlesh,<br>> <br>> Please provide a SIP trace (sip set debug on) of a successful u-law call. I'm<br>> especially interested in the dialog between the Asterisk server and the ITSP in<br>> this scenario.<br>> <br>> Also include the relevant sections of sip.conf and the dialplan.<br>> <br>> Regards,<br>> <br>> Matthew Roth<br>> InterMedia Marketing Solutions<br>> Software Engineer and Systems Developer<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div></body>
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