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It seems a firewall or signaling problem. The calling part is not
sending a ACK response to your host because it never get an "OK 200"
from your host.<br>
<br>
In other words, the called part is trying to send to the calling part <br>
a) TRYING 100, then<br>
b) RING 180 and finally<br>
c) OK 200 <br>
<br>
but the calling part seems not being receiving no signals from
you host. <br>
As a result, your host has sent 4 times "SIP/2.0 200 OK"
(retransmissions) to the calling part but it never got an ACK from the
other end to establish the communication.<br>
Then, the link is destroyed.<br>
<br>
regards,<br>
<pre class="moz-signature" cols="72">--
==================================
Miguel Oyarzo
Senior [ Network | Systems Design ] Engineer
<a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/mikeaustralia">http://www.linkedin.com/in/mikeaustralia</a>
Linux User: # 483188 - counter.li.org
Melbourne, Australia
</pre>
<br>
<br>
On 11/11/2012 9:46 PM, Eric Kuhnke wrote:
<blockquote
cite="mid:CAB69EHh8t5_F1zH7vX4Z711quJQ99LToVSAcXbKH7DYD_BxhsA@mail.gmail.com"
type="cite">
<pre style="word-wrap: break-word; white-space: pre-wrap;">Hi all,
I'm trying to troubleshoot an issue with my SIP service. All outgoing
calls work normally. The following is a SIP debug log from Asterisk. The
test setup is as follows:
One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk
to my local FreePBX/Asterisk 11.0 server which is at 10.0.0.17.
The Yealink phone doesn't seem to have any problem placing outgoing calls
through the Asterisk server, which is registered to Diamondcard. I can
reach both the Asterisk server itself (for example to use voicemail) or
call any number on the PSTN. Likewise I have the server configured to pass
incoming DID calls for myDIDnumber to extension 10. Calls from the PSTN to
myDIDnumber ring the phone, including CID passing, and will connect a full
duplex audio call session. The problem is that the phone won't stay
connected longer than 13 to 17 seconds.
When the phone is manually configured to use my account and password on the
diamondcard servers directly, both incoming and outgoing calls work
normally, with RTP/UDP port 5060 traffic passing through my NAT without
trouble. I have made no special modifications to the NAT.
13 seconds after picking up an incoming call, the phone disconnects at the
same time as the log shows this:
[2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Retransmission timeout
reached on transmission <a moz-do-not-send="true"
href="mailto:08a728706baea3b740aa806e41e9d13d@69.71.222.196">08a728706baea3b740aa806e41e9d13d@69.71.222.196</a> for
seqno 103 (Critical Response) -- See
<a moz-do-not-send="true"
href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>
Packet timed out after 17853ms with no response
[2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Hanging up call
<a moz-do-not-send="true"
href="mailto:08a728706baea3b740aa806e41e9d13d@69.71.222.196">08a728706baea3b740aa806e41e9d13d@69.71.222.196</a> - no reply to our critical
packet (see <a moz-do-not-send="true"
href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).
</pre>
<pre style="word-wrap: break-word; white-space: pre-wrap;">
</pre>
<pre style="word-wrap: break-word; white-space: pre-wrap;"><b>The full log and configuration is at:</b></pre>
<pre style="word-wrap: break-word;"><span
style="white-space: pre-wrap;"><b><a moz-do-not-send="true"
href="http://pastebin.com/1Mgn72vN">http://pastebin.com/1Mgn72vN</a></b></span></pre>
<pre wrap="">
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