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I have remote access to the server so I checked the canreinvite .. they
are all set to no. I can't try the call from here, I will get back to
you.<br>
<pre class="moz-signature" cols="72">Gary Baribault</pre>
<br>
On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote:
<blockquote
cite="mid:AANLkTin2ZOoWFg-rK4nKr79ersb91uX1k22f6VLl_y_n@mail.gmail.com"
type="cite">
<p>Do you agree something is blocking the audio in one direction? Can
you do a 'rtp debug' and then initiate a SIP call and see if there is
two way audio traffic. Also make sure these extensions have
'canreinvite=no'.<br>
</p>
<p>Zeeshan A Zakaria</p>
<p>--<br>
Sent from my Android phone with K-9 Mail.</p>
<blockquote type="cite">On 2010-06-01 7:02 PM, "Gary Baribault" <<a
moz-do-not-send="true" href="mailto:gary@baribault.net">gary@baribault.net</a>>
wrote:<br>
<br>
<div bgcolor="#ffffff" text="#000000">As I stated, the incoming
calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.<br>
<font color="#888888"><br>
Gary Baribault</font>
<p><font color="#500050"><br>
<br>
On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:<br>
><br>
> Output of 'iptables -L -n' would also be helpfu...</font></p>
</div>
<br>
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