I am running PJPROJECT 2.3 and Asterisk 13.0.0. I answer the call, about 15 seconds later, vitality hangs up on my cell phone. However, Asterisk is never notified When the OK (for the answer) occurs, the ACK seems to never be accepted. The OK recvd with ACK sent occurs several times.
Here are the pjsip.conf settings... [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net] type = aor remove_existing = yes qualify_frequency = 60 contact = sip:64.2.142.93 [outbound.vitelity.net] type = endpoint context = TestApp transport = transport1 aors = outbound.vitelity.net dtmf_mode = rfc4733 force_rport = yes rtp_symmetric = yes rewrite_contact = yes send_rpid = yes trust_id_inbound = yes disallow = all allow = ulaw direct_media = no [outbound.vitelity.net] type = identify endpoint = outbound.vitelity.net match = 64.2.142.93 When I Originate a call via AMI... Action: Originate ActionID: S8 Channel: PJSIP/[email protected]<mailto:PJSIP/[email protected]> Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable: CALLERID(num-pres)=allowed_passed_screen Async: true The call goes through to my cell phone. I answer on my cell phone and Asterisk sees the call being answered. However, Vitelity disconnects the cell phone about 15 seconds later. When looking at the PJSIP trace, the ACK repsonse to the 200 OK (Answer) are missing the Contact header. From what I understand that is likely the reason Vitelity doesn't seem to process the ACK. *CLI> -- Called [email protected]<mailto:[email protected]> <--- Transmitting SIP request (1018 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.11.166:5060;rport;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2 From: "John Doe" <sip:[email protected]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03 To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125 CSeq: 10207 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: "John Doe" <sip:[email protected]>;privacy=off;screen=no Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Type: application/sdp Content-Length: 241 v=0 o=- 2134048799 2134048799 IN IP4 192.168.11.166 s=Asterisk c=IN IP4 192.168.11.166 t=0 0 m=audio 16262 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 (384 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.11.166:5060;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2 From: "John Doe" <sip:[email protected]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03 To: <sip:[email protected]> Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125 CSeq: 10207 INVITE Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)) Content-Length: 0 <--- Received SIP response (851 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.11.166:5060;received=192.168.11.166;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2 Record-Route: <sip:64.2.142.93;lr=on> From: "John Doe" <sip:[email protected]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03 To: <sip:[email protected]>;tag=as466c6135 Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125 CSeq: 10207 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 240 v=0 o=root 3457 3457 IN IP4 66.241.99.145 s=session c=IN IP4 66.241.99.145 t=0 0 m=audio 11872 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 -- PJSIP/outbound.vitelity.net-00000000 is making progress > 0x60fc840 -- Probation passed - setting RTP source address to 66.241.99.145:11872 <--- Received SIP response (837 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.166:5060;received=192.168.11.166;rport=5060;branch=z9hG4bKPjc0fc94dc-bde3-441c-8a58-cd71e2d326f2 Record-Route: <sip:64.2.142.93;lr=on> From: "John Doe" <sip:[email protected]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03 To: <sip:[email protected]>;tag=as466c6135 Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125 CSeq: 10207 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 240 v=0 o=root 3457 3458 IN IP4 66.241.99.145 s=session c=IN IP4 66.241.99.145 t=0 0 m=audio 11872 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 (443 bytes) to UDP:64.2.142.93:5060 ---> ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.166:5060;rport;branch=z9hG4bKPj5fa1c45c-fb91-4d87-aa6b-9ae451dcd211 From: "John Doe" <sip:[email protected]>;tag=340be959-af87-4b77-a7a7-8aaf83940b03 To: <sip:[email protected]>;tag=as466c6135 Call-ID: e443f2a6-8f80-4493-b12b-22c48247c125 CSeq: 10207 ACK Route: <sip:64.2.142.93;lr> Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Length: 0 -- PJSIP/outbound.vitelity.net-00000000 answered -- Executing [createcall@TestApp:1] Set("PJSIP/outbound.vitelity.net-00000000", "EXTIVR=") in new stack -- Executing [createcall@TestApp:2] AGI("PJSIP/outbound.vitelity.net-00000000", "agi:async") in new stack
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