[SR-Users] Asterisk 1.8 and Kamailio 1.5 issue

Rowell Rufino rowiebiz at gmail.com
Wed Oct 26 20:25:35 CEST 2011


 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 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


 Regards,

Rowell
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