[SR-Users] Asterisk 1.8 and Kamailio 1.5 issue

Thu Oct 27 18:58:27 CEST 2011

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 at ser.gowireless.net 
> <mailto:sip%3A91421 at ser.gowireless.net>>;tag=6wkdms1r20
> To: <sip:9513261429 at ser.gowireless.net 
> <mailto:sip%3A9513261429 at 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 at 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 at ser.gowireless.net 
> <mailto:sip%3A91421 at ser.gowireless.net>>;tag=6wkdms1r20
> t: <sip:9513261429 at ser.gowireless.net 
> <mailto:sip%3A9513261429 at ser.gowireless.net>;user=phone>;tag=as0b87218f
> i: 3c26755bf15c-9iq08xqqblo6
> CSeq: 4 ACK
> Max-Forwards: 70
> m: <sip:91421 at 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 at ser.gowireless.net 
> <mailto:sip%3A91421 at ser.gowireless.net>>;tag=qi3i8ze6z8
> To: <sip:9513261429 at ser.gowireless.net 
> <mailto:sip%3A9513261429 at 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 at 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 at 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 at ser.gowireless.net 
> <mailto:sip%3A91421 at ser.gowireless.net>>;tag=qi3i8ze6z8
> t: <sip:9513261429 at ser.gowireless.net 
> <mailto:sip%3A9513261429 at ser.gowireless.net>;user=phone>;tag=as3f8c0f96
> i: 3c2676547a8d-2t5yi6jok1sv
> CSeq: 2 ACK
> Max-Forwards: 70
> m: <sip:91421 at 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 at 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 at ser.gowireless.net 
> <mailto:sip%3A9513261429 at ser.gowireless.net>;user=phone>;tag=as3f8c0f96
> To: "Virgil Menendez" <sip:91421 at ser.gowireless.net 
> <mailto:sip%3A91421 at 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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-- 
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

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