[SR-Users] Kamailio 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)

Daniel-Constantin Mierla miconda at gmail.com
Tue Nov 8 10:55:46 CET 2011


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 at ser.gowireless.net>;tag=6wkdms1r20
> To:<sip:9513261429 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 at 26/10/2011 10:22:41:481 (387 bytes):
>
> ACK sip:vm9513261429 at 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>;tag=6wkdms1r20
> t:<sip:9513261429 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>;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 at ser.gowireless.net>;tag=qi3i8ze6z8
> To:<sip:9513261429 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>
> 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 at 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>;tag=qi3i8ze6z8
> t:<sip:9513261429 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>;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 at 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;user=phone>;tag=as3f8c0f96
> To: "Virgil Menendez"<sip:91421 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
>
>
>
>
>

-- 
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




More information about the sr-users mailing list