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
--
Daniel-Constantin Mierla --
http://www.asipto.com
Kamailio Advanced Training, Dec 5-8, Berlin:
http://asipto.com/u/kat
http://linkedin.com/in/miconda --
http://twitter.com/miconda