Hello,
you have to provide the sip trace taken on the sip server, in order to
see what is received and what is sent out by kamailio. Looks like the
one you pasted here is from client.
You can use ngrep on kamailio server:
ngrep -d any -qt -W byline port 5060
Also, the packets you pasted next are from two different calls (see the
call-id header). The second seems to be completed ok, but something is
not good for asterisk and it sends bye. Maybe you can spot something in
the logs of asterisk.
Cheers,
Daniel
On 10/26/11 8:25 PM, Rowell Rufino wrote:
Hi,
We are having issues where the "OK" or "ACK" is that is coming from
the phone is not relayed by OpenSER to Asterisk.
Below is the sip trace... I am also attaching a tcpdump. Please help
what we can do.
Received from udp:10.1.10.80:5060 <http://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
<mailto:sip%3A91421@ser.gowireless.net>>;tag=6wkdms1r20
To: <sip:9513261429@ser.gowireless.net
<mailto:sip%3A9513261429@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 <http://10.1.10.80:5060> at 26/10/2011
10:22:41:481 (387 bytes):
ACK sip:vm9513261429@10.1.10.83:5060
<http://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
<mailto:sip%3A91421@ser.gowireless.net>>;tag=6wkdms1r20
t: <sip:9513261429@ser.gowireless.net
<mailto:sip%3A9513261429@ser.gowireless.net>;user=phone>;tag=as0b87218f
i: 3c26755bf15c-9iq08xqqblo6
CSeq: 4 ACK
Max-Forwards: 70
m: <sip:91421@10.30.0.64:5060 <http://sip:91421@10.30.0.64:5060>>;reg-id=1
l: 0
------------------------------------------------------------------------
Received from udp:10.1.10.80:5060 <http://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
<mailto:sip%3A91421@ser.gowireless.net>>;tag=qi3i8ze6z8
To: <sip:9513261429@ser.gowireless.net
<mailto:sip%3A9513261429@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
<http://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 <http://10.1.10.80:5060> at 26/10/2011
10:22:42:132 (385 bytes):
ACK sip:9513261429@10.1.10.83:5060
<http://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
<mailto:sip%3A91421@ser.gowireless.net>>;tag=qi3i8ze6z8
t: <sip:9513261429@ser.gowireless.net
<mailto:sip%3A9513261429@ser.gowireless.net>;user=phone>;tag=as3f8c0f96
i: 3c2676547a8d-2t5yi6jok1sv
CSeq: 2 ACK
Max-Forwards: 70
m: <sip:91421@10.30.0.64:5060 <http://sip:91421@10.30.0.64:5060>>;reg-id=1
l: 0
------------------------------------------------------------------------
Received from udp:10.1.10.80:5060 <http://10.1.10.80:5060> at
26/10/2011 10:22:42:232 (503 bytes):
BYE sip:91421@10.30.0.64:5060 <http://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
<mailto:sip%3A9513261429@ser.gowireless.net>;user=phone>;tag=as3f8c0f96
To: "Virgil Menendez" <sip:91421@ser.gowireless.net
<mailto:sip%3A91421@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
Regards,
Rowell
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