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

[reeju jain]

Hi,

I am also facing the same issue.Kindly suggest me have you resolved this
issue?

Regards:
Reeju jain
9711071741


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
> 
> 
> 
> 
> 
> 

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