Hello, I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM URI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIP servers (it doesn't recognize the response / request). Below you can see an incoming INVITE and the outgoing 200OK response. I have highlighted the issue in Yellow. Does anyone know of a solution / workaround for this issue?
<--- Received SIP request (648 bytes) from UDP:172.16.60.160:5061 ---> INVITE sip:[email protected]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK-29450-3-0 Max-Forwards: 60 From: <sip:[email protected]:5061;user=phone>;tag=3 To: <sip:[email protected]:5060;user=phone> Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:10.1.1.1:5060> User-Agent: Simulator Supported: 100rel Privacy: id Min-SE: 90 Content-Type: application/sdp Content-Length: 201 v=0 o=172.16.60.160 10864 2 IN IP4 172.16.60.160 s=SIP Call c=IN IP4 172.16.60.160 t=0 0 a=sendrecv m=audio 60000 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <--- Transmitting SIP response (730 bytes) to UDP:172.16.60.160:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1:5060 ;rport;received=172.16.60.160;branch=z9hG4bK-29450-3-0 Call-ID: [email protected] From: <sip:[email protected];user=phone>;tag=3 To: <sip:[email protected] ;user=phone>;tag=4f7ef94f-fb15-4bf5-94bd-4e43fe-299655 CSeq: 1 INVITE Contact: <sip:172.16.60.160:5060> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 189 v=0 o=- 10864 4 IN IP4 10.2.0.67 s=Asterisk c=IN IP4 172.16.60.160 t=0 0 m=audio 19404 RTP/AVP 8 c=IN IP4 172.16.60.160 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:150 a=sendrecv
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