Hi! I also have another problem: since opensips is configured as asterisk peer is *'s config, the BYE asterisk sends does not get to the load balanced UA when it is NATed. Obviously, this is because there is no Via header in the BYE message with received and rport fields (sip trace below). Is there any way to solve this? Somehow remember the Via header for this call and add it to BYE when it passes through opensips (and matches the call)? Here is the trace from *'s point of view:
<--- SIP read from xxx.yyy.zzz.90:5060 ---> INVITE sip:t...@xxx.yyy.zzz.90 SIP/2.0 Record-Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963> Via: SIP/2.0/UDP xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.0 Via: SIP/2.0/UDP 192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKhkaqxvra Max-Forwards: 69 Proxy-Authorization: Digest username="from",realm="asterisk",nonce="51e71e69",uri="sip:t...@xxx.yyy.zzz.90",response="c643612 2f92fe7cf08006dc154705638",algorithm=MD5 To: <sip:t...@xxx.yyy.zzz.90> From: "aaa" <sip:f...@xxx.yyy.zzz.90>;tag=dwage Call-ID: vpdjrkrfbsmj...@aalexandrov.site CSeq: 131 INVITE Contact: <sip:f...@192.168.128.205> Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.4.2 Content-Length: 313 v=0 o=twinkle 830887642 558243211 IN IP4 192.168.128.205 s=- c=IN IP4 192.168.128.205 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <--- Transmitting (NAT) to xxx.yyy.zzz.90:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.0;received=xxx.yyy.zzz.90 Via: SIP/2.0/UDP 192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKhkaqxvra Record-Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963> From: "aaa" <sip:f...@xxx.yyy.zzz.90>;tag=dwage To: <sip:t...@xxx.yyy.zzz.90> Call-ID: vpdjrkrfbsmj...@aalexandrov.site CSeq: 131 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:t...@xxx.yyy.zzz.94> Content-Length: 0 <--- Reliably Transmitting (NAT) to xxx.yyy.zzz.90:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.0;received=xxx.yyy.zzz.90 Via: SIP/2.0/UDP 192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKhkaqxvra Record-Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963> From: "aaa" <sip:f...@xxx.yyy.zzz.90>;tag=dwage To: <sip:t...@xxx.yyy.zzz.90>;tag=as671f3653 Call-ID: vpdjrkrfbsmj...@aalexandrov.site CSeq: 131 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:t...@xxx.yyy.zzz.94> Content-Type: application/sdp Content-Length: 264 v=0 o=root 29029 29029 IN IP4 xxx.yyy.zzz.94 s=session c=IN IP4 xxx.yyy.zzz.94 t=0 0 m=audio 15576 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- SIP read from xxx.yyy.zzz.90:5060 ---> ACK sip:t...@xxx.yyy.zzz.94 SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.2 Via: SIP/2.0/UDP 192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKgvysatjo Max-Forwards: 69 Proxy-Authorization: Digest username="from",realm="asterisk",nonce="51e71e69",uri="sip:t...@xxx.yyy.zzz.90",response="c6436122f92fe7cf08006dc154705638",algorithm=MD5 To: <sip:t...@xxx.yyy.zzz.90>;tag=as671f3653 From: "aaa" <sip:f...@xxx.yyy.zzz.90>;tag=dwage Call-ID: vpdjrkrfbsmj...@aalexandrov.site CSeq: 131 ACK User-Agent: Twinkle/1.4.2 Content-Length: 0 Reliably Transmitting (NAT) to xxx.yyy.zzz.90:5060: BYE sip:f...@192.168.128.205 SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.zzz.94:5060;branch=z9hG4bK1890cc40;rport Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963> From: <sip:t...@xxx.yyy.zzz.90>;tag=as671f3653 To: "aaa" <sip:f...@xxx.yyy.zzz.90>;tag=dwage Call-ID: vpdjrkrfbsmj...@aalexandrov.site CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 Retransmitting #1 (NAT) to xxx.yyy.zzz.90:5060: BYE sip:f...@192.168.128.205 SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.zzz.94:5060;branch=z9hG4bK1890cc40;rport Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963> From: <sip:t...@xxx.yyy.zzz.90>;tag=as671f3653 To: "aaa" <sip:f...@xxx.yyy.zzz.90>;tag=dwage Call-ID: vpdjrkrfbsmj...@aalexandrov.site CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
