Hi, I am also facing the same issue.Kindly suggest me have you resolved this issue?
Regards: Reeju jain 9711071741 Rowie wrote: > > Hi, > > We are having an issue where a phone (snom in particular) cannot make a > call through Asterisk. It just hangup and does not allow the call to go > through. I am including a a sip trace on this thread to show what is > happening within the call. Please see below: > > Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): > > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP > 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 > From: "Virgil Menendez" <sip:91...@ser.gowireless.net>;tag=6wkdms1r20 > To: <sip:9513261...@ser.gowireless.net;user=phone>;tag=as0b87218f > Call-ID: 3c26755bf15c-9iq08xqqblo6 > CSeq: 4 INVITE > Server: Asterisk PBX 1.8.7.1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Content-Length: 0 > > > > > -------------------------------------------------------------------------------- > > Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): > > ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 > v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport > Route: <sip:10.1.10.80;lr=on> > f: "Virgil Menendez" <sip:91...@ser.gowireless.net>;tag=6wkdms1r20 > t: <sip:9513261...@ser.gowireless.net;user=phone>;tag=as0b87218f > i: 3c26755bf15c-9iq08xqqblo6 > CSeq: 4 ACK > Max-Forwards: 70 > m: <sip:91421@10.30.0.64:5060>;reg-id=1 > l: 0 > > > > > -------------------------------------------------------------------------------- > > Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): > > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 > Record-Route: <sip:10.1.10.80;lr=on> > From: "Virgil Menendez" <sip:91...@ser.gowireless.net>;tag=qi3i8ze6z8 > To: <sip:9513261...@ser.gowireless.net;user=phone>;tag=as3f8c0f96 > Call-ID: 3c2676547a8d-2t5yi6jok1sv > CSeq: 2 INVITE > Server: Asterisk PBX 1.8.7.1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Session-Expires: 1800;refresher=uas > Contact: <sip:9513261429@10.1.10.83:5060> > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=root 1355451627 1355451627 IN IP4 10.1.10.83 > s=Asterisk PBX 1.8.7.1 > c=IN IP4 10.1.10.83 > t=0 0 > m=audio 16094 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > > > -------------------------------------------------------------------------------- > > Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): > > ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 > v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport > Route: <sip:10.1.10.80;lr=on> > f: "Virgil Menendez" <sip:91...@ser.gowireless.net>;tag=qi3i8ze6z8 > t: <sip:9513261...@ser.gowireless.net;user=phone>;tag=as3f8c0f96 > i: 3c2676547a8d-2t5yi6jok1sv > CSeq: 2 ACK > Max-Forwards: 70 > m: <sip:91421@10.30.0.64:5060>;reg-id=1 > l: 0 > > > > > -------------------------------------------------------------------------------- > > Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): > > BYE sip:91421@10.30.0.64:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 > Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 > Max-Forwards: 69 > From: <sip:9513261...@ser.gowireless.net;user=phone>;tag=as3f8c0f96 > To: "Virgil Menendez" <sip:91...@ser.gowireless.net>;tag=qi3i8ze6z8 > Call-ID: 3c2676547a8d-2t5yi6jok1sv > CSeq: 102 BYE > User-Agent: Asterisk PBX 1.8.7.1 > X-Asterisk-HangupCause: Protocol error, unspecified > X-Asterisk-HangupCauseCode: 111 > Content-Length: 0 > > > > > > -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error%2C-Unspecified%29-tp32792297p35060005.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
