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Hello,<br>
<br>
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.<br>
<br>
You can use ngrep on kamailio server:<br>
<br>
ngrep -d any -qt -W byline port 5060<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
On 10/26/11 8:25 PM, Rowell Rufino wrote:
<blockquote
cite="mid:CABeSUtNi5xX9rq7O1+nn2YdHc8jeDhcpB-VEQWz7dNXOLgDzLw@mail.gmail.com"
type="cite">
<div style="FONT-FAMILY: Tahoma; DIRECTION: ltr; COLOR: #000000;
FONT-SIZE: x-small">
<div dir="ltr"><font color="#000000" face="Tahoma" size="2">Hi,</font></div>
<div dir="ltr"> </div>
<div dir="ltr"><font face="tahoma">We are having issues where
the "OK" or "ACK" is that is coming from the phone is not
relayed by OpenSER to Asterisk.</font></div>
<div dir="ltr"> </div>
<div dir="ltr"><font face="tahoma">Below is the sip trace... I
am also attaching a tcpdump. Please help what we can do. </font></div>
<div dir="ltr"> </div>
<div dir="ltr"> </div>
<div dir="ltr"><font face="tahoma">
<p>Received from udp:<a moz-do-not-send="true"
href="http://10.1.10.80:5060">10.1.10.80:5060</a> at
26/10/2011 10:22:41:476 (490 bytes):</p>
<p class="text4">SIP/2.0 481 Call/Transaction Does Not Exist<br>
Via: SIP/2.0/UDP
10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060<br>
From: "Virgil Menendez" <<a moz-do-not-send="true"
href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=6wkdms1r20<br>
To: <<a moz-do-not-send="true"
href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as0b87218f<br>
Call-ID: 3c26755bf15c-9iq08xqqblo6<br>
CSeq: 4 INVITE<br>
Server: Asterisk PBX 1.8.7.1<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>
Content-Length: 0<br>
<br>
</p>
<hr>
<p>Sent to udp:<a moz-do-not-send="true"
href="http://10.1.10.80:5060">10.1.10.80:5060</a> at
26/10/2011 10:22:41:481 (387 bytes):</p>
<p class="text4">ACK <a moz-do-not-send="true"
href="http://sip:vm9513261429@10.1.10.83:5060">sip:vm9513261429@10.1.10.83:5060</a>
SIP/2.0<br>
v: SIP/2.0/UDP
10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport<br>
Route: <a class="moz-txt-link-rfc2396E" href="sip:10.1.10.80;lr=on"><sip:10.1.10.80;lr=on></a><br>
f: "Virgil Menendez" <<a moz-do-not-send="true"
href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=6wkdms1r20<br>
t: <<a moz-do-not-send="true"
href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as0b87218f<br>
i: 3c26755bf15c-9iq08xqqblo6<br>
CSeq: 4 ACK<br>
Max-Forwards: 70<br>
m: <<a moz-do-not-send="true"
href="http://sip:91421@10.30.0.64:5060">sip:91421@10.30.0.64:5060</a>>;reg-id=1<br>
l: 0<br>
<br>
</p>
<hr>
<p>Received from udp:<a moz-do-not-send="true"
href="http://10.1.10.80:5060">10.1.10.80:5060</a> at
26/10/2011 10:22:42:130 (868 bytes):</p>
<p class="text4">SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP
10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060<br>
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:10.1.10.80;lr=on"><sip:10.1.10.80;lr=on></a><br>
From: "Virgil Menendez" <<a moz-do-not-send="true"
href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=qi3i8ze6z8<br>
To: <<a moz-do-not-send="true"
href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as3f8c0f96<br>
Call-ID: 3c2676547a8d-2t5yi6jok1sv<br>
CSeq: 2 INVITE<br>
Server: Asterisk PBX 1.8.7.1<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>
Session-Expires: 1800;refresher=uas<br>
Contact: <<a moz-do-not-send="true"
href="http://sip:9513261429@10.1.10.83:5060">sip:9513261429@10.1.10.83:5060</a>><br>
Content-Type: application/sdp<br>
Content-Length: 256<br>
<br>
v=0<br>
o=root 1355451627 1355451627 IN IP4 10.1.10.83<br>
s=Asterisk PBX 1.8.7.1<br>
c=IN IP4 10.1.10.83<br>
t=0 0<br>
m=audio 16094 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
</p>
<hr>
<p>Sent to udp:<a moz-do-not-send="true"
href="http://10.1.10.80:5060">10.1.10.80:5060</a> at
26/10/2011 10:22:42:132 (385 bytes):</p>
<p class="text4">ACK <a moz-do-not-send="true"
href="http://sip:9513261429@10.1.10.83:5060">sip:9513261429@10.1.10.83:5060</a>
SIP/2.0<br>
v: SIP/2.0/UDP
10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport<br>
Route: <a class="moz-txt-link-rfc2396E" href="sip:10.1.10.80;lr=on"><sip:10.1.10.80;lr=on></a><br>
f: "Virgil Menendez" <<a moz-do-not-send="true"
href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=qi3i8ze6z8<br>
t: <<a moz-do-not-send="true"
href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as3f8c0f96<br>
i: 3c2676547a8d-2t5yi6jok1sv<br>
CSeq: 2 ACK<br>
Max-Forwards: 70<br>
m: <<a moz-do-not-send="true"
href="http://sip:91421@10.30.0.64:5060">sip:91421@10.30.0.64:5060</a>>;reg-id=1<br>
l: 0<br>
<br>
</p>
<hr>
<p>Received from udp:<a moz-do-not-send="true"
href="http://10.1.10.80:5060">10.1.10.80:5060</a> at
26/10/2011 10:22:42:232 (503 bytes):</p>
<p class="text4">BYE <a moz-do-not-send="true"
href="http://sip:91421@10.30.0.64:5060">sip:91421@10.30.0.64:5060</a>
SIP/2.0<br>
Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0<br>
Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1<br>
Max-Forwards: 69<br>
From: <<a moz-do-not-send="true"
href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as3f8c0f96<br>
To: "Virgil Menendez" <<a moz-do-not-send="true"
href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=qi3i8ze6z8<br>
Call-ID: 3c2676547a8d-2t5yi6jok1sv<br>
CSeq: 102 BYE<br>
User-Agent: Asterisk PBX 1.8.7.1<br>
<strong>X-Asterisk-HangupCause: Protocol error,
unspecified<br>
</strong>X-Asterisk-HangupCauseCode: 111<br>
Content-Length: 0<br>
</p>
</font></div>
<div dir="ltr"> </div>
<div dir="ltr"> </div>
<div class="BodyFragment"><font size="2">
<div class="PlainText">Regards,<br>
<br>
Rowell</div>
</font></div>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a></pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
Kamailio Advanced Training, Dec 5-8, Berlin: <a class="moz-txt-link-freetext" href="http://asipto.com/u/kat">http://asipto.com/u/kat</a>
<a class="moz-txt-link-freetext" href="http://linkedin.com/in/miconda">http://linkedin.com/in/miconda</a> -- <a class="moz-txt-link-freetext" href="http://twitter.com/miconda">http://twitter.com/miconda</a></pre>
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