Please don't beat me up on unsupported phones. Sometimes, we have to run with the environment not dictate it.
We are able to get a sip extension to register and receive inbound calls. Outbound calls from the extension appear to be rejected by the pbx. I am looking for some expert recommendations on what we have going wrong in the invite. We have attempted several different ways and are somewhat just flailing around now.
Linux V20 Update 1 Build 731
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:100@[redacted ip] SIP/2.0 Via: SIP/2.0/UDP [redacted ip];branch=z9hG4bK-3135036560;rport=39379 Max-Forwards: 70 Contact: <sip:100@[redacted ip]> To: "Asterisk"<sip:[email protected]> From: "Asterisk"<sip:[email protected]>;tag=3033313236636336313363340132313034383431373238 Call-ID: 274345777184968610172684 CSeq: 1 INVITE Accept: application/sdp User-Agent: PBX Content-Length: 0
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:2604692170@[redacted 3cx domain] SIP/2.0 Via: SIP/2.0/UDP [redacted ip]:5060;branch=z9hG4bKtsmgih1070dl1ccpsh10.1 Max-Forwards: 69 Route: <sip:99892604692170@[redacted 3cx domain]:5060;lr> Contact: <sip:600@[redacted IP]:5060;transport=udp> To: <sip:2604692170@[redacted 3cx domain]:5060> From: "redacted name" <sip:8088@[redacted ip]:5060>;tag=8dbeb758-e26d-41e8-957f-bf82459b0ffc Call-ID: 7fd2c4dd-144f-45d1-9585-7b696fdb783c CSeq: 8588 INVITE Session-Expires: 1800 Min-SE: 90 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Content-Type: application/sdp Supported: 100rel, timer, replaces, norefersub, histinfo User-Agent: redacted namePhone Integration Gateway Content-Length: 341 v=0 o=- 1899558860 1899558860 IN IP4 [redacted ip] s=Asterisk c=IN IP4 [redacted ip] t=0 0 m=audio 52860 RTP/AVP 0 8 3 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
We are able to get a sip extension to register and receive inbound calls. Outbound calls from the extension appear to be rejected by the pbx. I am looking for some expert recommendations on what we have going wrong in the invite. We have attempted several different ways and are somewhat just flailing around now.
Linux V20 Update 1 Build 731
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:100@[redacted ip] SIP/2.0 Via: SIP/2.0/UDP [redacted ip];branch=z9hG4bK-3135036560;rport=39379 Max-Forwards: 70 Contact: <sip:100@[redacted ip]> To: "Asterisk"<sip:[email protected]> From: "Asterisk"<sip:[email protected]>;tag=3033313236636336313363340132313034383431373238 Call-ID: 274345777184968610172684 CSeq: 1 INVITE Accept: application/sdp User-Agent: PBX Content-Length: 0
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:2604692170@[redacted 3cx domain] SIP/2.0 Via: SIP/2.0/UDP [redacted ip]:5060;branch=z9hG4bKtsmgih1070dl1ccpsh10.1 Max-Forwards: 69 Route: <sip:99892604692170@[redacted 3cx domain]:5060;lr> Contact: <sip:600@[redacted IP]:5060;transport=udp> To: <sip:2604692170@[redacted 3cx domain]:5060> From: "redacted name" <sip:8088@[redacted ip]:5060>;tag=8dbeb758-e26d-41e8-957f-bf82459b0ffc Call-ID: 7fd2c4dd-144f-45d1-9585-7b696fdb783c CSeq: 8588 INVITE Session-Expires: 1800 Min-SE: 90 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Content-Type: application/sdp Supported: 100rel, timer, replaces, norefersub, histinfo User-Agent: redacted namePhone Integration Gateway Content-Length: 341 v=0 o=- 1899558860 1899558860 IN IP4 [redacted ip] s=Asterisk c=IN IP4 [redacted ip] t=0 0 m=audio 52860 RTP/AVP 0 8 3 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
