I am not sure what you mean by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK
---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555@64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93> Contact: <sip:15062fef-986e-4fcf-a93e-06b28da02...@xxx.xxx.xxx.xxx:5060> Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, REGISTER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Remote-Party-ID: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;privacy=off;screen=no Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Type: application/sdp Content-Length: 239 v=0 o=- 540555224 540555224 IN IP4 XXX.XXX.XXX.XXX s=Asterisk c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 10030 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (378 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93> Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 INVITE Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)) Content-Length: 0 <--- Received SIP response (844 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a Record-Route: <sip:64.2.142.93;lr=on> From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:18005555555@64.2.142.192> Content-Type: application/sdp Content-Length: 240 v=0 o=root 32312 32312 IN IP4 64.2.142.192 s=session c=IN IP4 64.2.142.192 t=0 0 m=audio 17494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Phone is ringing. Next, I answer my cell phone <--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a Record-Route: <sip:64.2.142.93;lr=on> From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:18005555555@64.2.142.192> Content-Type: application/sdp Content-Length: 240 v=0 o=root 32312 32313 IN IP4 64.2.142.192 s=session c=IN IP4 64.2.142.192 t=0 0 m=audio 17494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 ---> ACK sip:18005555555@64.2.142.93:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 ACK Route: <sip:64.2.142.93;lr> Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 <--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a Record-Route: <sip:64.2.142.93;lr=on> From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:18005555555@64.2.142.192> Content-Type: application/sdp Content-Length: 240 v=0 o=root 32312 32313 IN IP4 64.2.142.192 s=session c=IN IP4 64.2.142.192 t=0 0 m=audio 17494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 ---> ACK sip:18005555555@64.2.142.93:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 ACK Route: <sip:64.2.142.93;lr> Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 <--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a Record-Route: <sip:64.2.142.93;lr=on> From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:18005555555@64.2.142.192> Content-Type: application/sdp Content-Length: 240 v=0 o=root 32312 32313 IN IP4 64.2.142.192 s=session c=IN IP4 64.2.142.192 t=0 0 m=audio 17494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 ---> ACK sip:18005555555@64.2.142.93:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 ACK Route: <sip:64.2.142.93;lr> Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 <--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a Record-Route: <sip:64.2.142.93;lr=on> From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:18005555555@64.2.142.192> Content-Type: application/sdp Content-Length: 240 v=0 o=root 32312 32313 IN IP4 64.2.142.192 s=session c=IN IP4 64.2.142.192 t=0 0 m=audio 17494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 ---> ACK sip:18005555555@64.2.142.93:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23191 ACK Route: <sip:64.2.142.93;lr> Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 At this point, my cell phone is disconnected, but Asterisk still thinks there is a call. Next I issue a hangup to Asterisk and it terminates the call <--- Transmitting SIP request (456 bytes) to UDP:64.2.142.93:5060 ---> BYE sip:18005555555@64.2.142.192 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPjd458d550-7bce-4008-813c-c84a0e446a86 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23192 BYE Route: <sip:64.2.142.93;lr> Reason: Q.850;cause=0 Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 <--- Received SIP response (507 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPjd458d550-7bce-4008-813c-c84a0e446a86 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555@64.2.142.93>;tag=as7aad862a Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43 CSeq: 23192 BYE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <--- Transmitting SIP request (473 bytes) to UDP:64.2.142.93:5060 ---> OPTIONS sip:64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPjddc1c054-48ce-4714-ba4e-9690aa9be55b From: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa...@xxx.xxx.xxx.xxx>;tag=9c4172da-1b32-442b-bea0-75ca3530b661 To: <sip:64.2.142.93> Contact: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa...@xxx.xxx.xxx.xxx:5060> Call-ID: 0817263d-519b-4bcc-aa11-a5bbd8c21f2f CSeq: 20166 OPTIONS Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 <--- Transmitting SIP request (487 bytes) to UDP:192.168.10.235:5060 ---> OPTIONS sip:291@192.168.10.235 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj46636dcf-d468-4738-a053-22334fe4523b From: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7...@xxx.xxx.xxx.xxx>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220 To: <sip:291@192.168.10.235> Contact: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7...@xxx.xxx.xxx.xxx:5060> Call-ID: 9664c7bb-b978-4443-9190-bba0d805be47 CSeq: 62000 OPTIONS Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 <--- Received SIP response (465 bytes) from UDP:192.168.10.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj46636dcf-d468-4738-a053-22334fe4523b To: <sip:291@192.168.10.235> From: <sip:99d5e64e-8f13-4904-9bbf-0a29865e7...@xxx.xxx.xxx.xxx>;tag=86edf61e-01fd-4fb8-8d38-60d2eabec220 Call-ID: 9664c7bb-b978-4443-9190-bba0d805be47 CSeq: 62000 OPTIONS Contact: <sip:Infinity@192.168.10.235> Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REGISTER,REFER,NOTIFY Supported: replaces Accept: application/sdp <--- Received SIP response (462 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 200 OPTIONS is almost as pointless as T38 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bKPjddc1c054-48ce-4714-ba4e-9690aa9be55b From: <sip:b5a0e71d-9216-4f3c-aedb-a48ecd2fa...@xxx.xxx.xxx.xxx>;tag=9c4172da-1b32-442b-bea0-75ca3530b661 To: <sip:64.2.142.93>;tag=37c906215f6623e2b0c0b8aa47fb6fb6.bc9b Call-ID: 0817263d-519b-4bcc-aa11-a5bbd8c21f2f CSeq: 20166 OPTIONS Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)) Content-Length: 0 --- PJSIP --- Reliably Transmitting (NAT) to 64.2.142.93:5060: INVITE sip:8005555555@64.2.142.93:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2f5e2a55;rport Max-Forwards: 70 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=as5678b23c To: <sip:8005555555@64.2.142.93:5060> Contact: <sip:2...@xxx.xxx.xxx.xxx:5060> Call-ID: 783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.2.0 Date: Wed, 10 Dec 2014 21:56:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 1537 v=0 o=root 133352036 133352036 IN IP4 XXX.XXX.XXX.XXX s=Asterisk PBX 12.2.0 c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 13752 RTP/AVP 10 4 3 0 8 111 5 7 18 110 117 97 112 9 118 102 115 116 119 107 96 108 109 113 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:118 L16/16000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:119 speex/32000 a=rtpmap:107 opus/48000/2 a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 a=rtpmap:96 SILK/8000 a=fmtp:96 maxaveragebitrate=10000 a=fmtp:96 usedtx=0 a=fmtp:96 useinbandfec=1 a=rtpmap:108 SILK/12000 a=fmtp:108 maxaveragebitrate=12000 a=fmtp:108 usedtx=0 a=fmtp:108 useinbandfec=1 a=rtpmap:109 SILK/16000 a=fmtp:109 maxaveragebitrate=20000 a=fmtp:109 usedtx=0 a=fmtp:109 useinbandfec=1 a=rtpmap:113 SILK/24000 a=fmtp:113 maxaveragebitrate=30000 a=fmtp:113 usedtx=0 a=fmtp:113 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:20 a=sendrecv --- <--- SIP read from UDP:64.2.142.93:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2f5e2a55;rport=5060 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=as5678b23c To: <sip:8005555555@64.2.142.93:5060> Call-ID: 783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060 CSeq: 102 INVITE Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:64.2.142.93:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bK2f5e2a55;rport=5060 Record-Route: <sip:64.2.142.93;lr=on> From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=as5678b23c To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2 Call-ID: 783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060 CSeq: 102 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:18005555555@66.241.99.161> Content-Type: application/sdp Content-Length: 312 v=0 o=root 15367 15367 IN IP4 66.241.99.161 s=session c=IN IP4 66.241.99.161 t=0 0 m=audio 11460 RTP/AVP 0 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 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 <-------------> --- (13 headers 15 lines) --- list_route: route/path hop: <sip:64.2.142.93;lr=on> Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 66.241.99.161:11460 <--- SIP read from UDP:64.2.142.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bK2f5e2a55;rport=5060 Record-Route: <sip:64.2.142.93;lr=on> From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=as5678b23c To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2 Call-ID: 783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060 CSeq: 102 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:18005555555@66.241.99.161> Content-Type: application/sdp Content-Length: 312 v=0 o=root 15367 15368 IN IP4 66.241.99.161 s=session c=IN IP4 66.241.99.161 t=0 0 m=audio 11460 RTP/AVP 0 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 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 <-------------> --- (13 headers 15 lines) --- Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 66.241.99.161:11460 list_route: route/path hop: <sip:64.2.142.93;lr=on> Transmitting (NAT) to 64.2.142.93:5060: ACK sip:18005555555@66.241.99.161 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ba6d973;rport Route: <sip:64.2.142.93;lr=on> Max-Forwards: 70 From: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=as5678b23c To: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2 Contact: <sip:2...@xxx.xxx.xxx.xxx:5060> Call-ID: 783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 12.2.0 Content-Length: 0 --- [Dec 10 21:56:25] NOTICE[3691][C-00000001]: channel.c:4163 __ast_read: Dropping incompatible voice frame on SIP/outbound.vitelity.net-00000001 of format ulaw since our native format has changed to (gsm) <--- SIP read from UDP:64.2.142.93:5060 ---> BYE sip:2...@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 64.2.142.93;branch=z9hG4bKd12e.ec461794.0 Via: SIP/2.0/UDP 66.241.99.161:5060;received=66.241.99.161;branch=z9hG4bK071dea37;rport=5060 From: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2 To: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=as5678b23c Call-ID: 783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060 CSeq: 102 BYE User-Agent: packetrino Max-Forwards: 69 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 64.2.142.93:5060 (NAT) Scheduling destruction of SIP dialog '783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 64.2.142.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.2.142.93;branch=z9hG4bKd12e.ec461794.0;received=64.2.142.93;rport=5060 Via: SIP/2.0/UDP 66.241.99.161:5060;received=66.241.99.161;branch=z9hG4bK071dea37;rport=5060 From: <sip:8005555555@64.2.142.93:5060>;tag=as2968a7d2 To: "Dan" <sip:2...@xxx.xxx.xxx.xxx>;tag=as5678b23c Call-ID: 783c897d153242595013ae516ebaf...@xxx.xxx.xxx.xxx:5060 CSeq: 102 BYE Server: Asterisk PBX 12.2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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