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

[Rowie]

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





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