Hello, I am experiencing strange behaviour from one specific provider. 80% of calls work properly, some of them fall into the validate_dialog condition, returncode = -1 i've checked INVITE and ACK message , i don't see anything that could cause that. fix_dialog_route seems to fix the problem but I would love to understand what is causing it. i noticed that with other providers when the dialog is not valid there is an ERROR message in the log files "ERROR:dialog:dlg_validate_dialog: failed to validate remote contact" with this one , there is no error. I compared the call with another working one from the same provider , I see no difference. Here are the initial invite, the 200 ok from opensips to provider, and the ACK that does not match during dialog validation.
proxyfqdn.example: my opensips FQDN pro.vid.der.ip : provider ip address my.pbx.behind.opensips.ip : my pbx behind opensips (asterisk box) INITIAL INVITE : INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP pro.vid.der.ip:5060;branch=z9hG4bK695c5157 Max-Forwards: 70 From: "anonymous" <sip:[email protected]>;tag=as6a0e2632 To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 18.3.0 Date: Mon, 17 Apr 2023 18:52:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-ADDTIONAL-UUID: <p(< Privacy: id P-Asserted-Identity: <sip:[email protected]:5060> XSHDATAID: T1798 Diversion: <sip:[email protected]>;reason=unconditional Content-Type: application/sdp Content-Length: 250 v=0 o=root 665874795 665874795 IN IP4 pro.vid.der.ip s=Asterisk PBX 18.3.0 c=IN IP4 pro.vid.der.ip t=0 0 m=audio 7162 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -------------------------------------- 200 OK (Proxy to provider) SIP/2.0 200 OK Via: SIP/2.0/UDP pro.vid.der.ip:5060;branch=z9hG4bK695c5157 Record-Route: <sip:pro.xy.ip.address:5060;lr;ftag=as6a0e2632;did=1bc.b66272e6> From: "anonymous" <sip:[email protected]>;tag=as6a0e2632 To: <sip:[email protected]>;tag=as2b2572fa Call-ID: [email protected]:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.11.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Type: application/sdp Require: timer Content-Length: 237 v=0 o=root 1475118129 1475118129 IN IP4 my.pbx.behind.opensips.ip s=MYMEDIASERVER c=IN IP4 my.pbx.behind.opensips.ip t=0 0 m=audio 17486 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv -------------------------------------------------------- ACK that cause dialog invalidation : ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP pro.vid.der.ip:5060;branch=z9hG4bK11c1df1f Route: <sip:pro.xy.ip.address:5060;lr;ftag=as6a0e2632;did=1bc.b66272e6> Max-Forwards: 70 From: "anonymous" <sip:[email protected]>;tag=as6a0e2632 To: <sip:[email protected]>;tag=as2b2572fa Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 18.3.0 Content-Length: 0
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