Hello,
the replies is coming from asterisk, so the issue is very likely to be there. Maybe you can run asterisk in debug mode and you get some log message indicating what is the problem.
On another hand, the trace you provided is not complete, in order to tell whether the forwarding in kamailio went ok, you need to get full sip trace from kamailio server, like with:
ngrep -d any -qt -W byline port 5060
Run it on kamailio server for such call, starting with the initial INVITE getting to kamailio, till at least the 481 reply.
Then we can see what was flowing through kamailio and if something looks wrong in the signaling packages.
Cheers, Daniel
On 11/7/11 11:55 PM, 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:91421@ser.gowireless.net;tag=6wkdms1r20 To:sip:9513261429@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:91421@ser.gowireless.net;tag=6wkdms1r20 t:sip:9513261429@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:91421@ser.gowireless.net;tag=qi3i8ze6z8 To:sip:9513261429@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:91421@ser.gowireless.net;tag=qi3i8ze6z8 t:sip:9513261429@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:9513261429@ser.gowireless.net;user=phone;tag=as3f8c0f96 To: "Virgil Menendez"sip:91421@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