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I have checked, the users have ulaw, then alaw, the phones are set to
711u then 711a which is the same thing (I think).<br>
<br>
Gary Baribault<br>
<br>
On 06/02/2010 08:32 AM, taimur hasan wrote:
<blockquote cite="mid:SNT137-w55272CD27B3EEC2DFC1B37E1D00@phx.gbl"
type="cite">
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Also check the codecs as if you are using g729 or g723, there is a
chance that they are not available in codecs directory (
/usr/lib/asterisk/modules).<br>
<p align="center"><font color="#000000"
face="Lucida Handwriting, Cursive" size="6"><strong>-<font
style="background-color: rgb(255, 255, 255);">THQ</font>- !!!ONE</strong></font></p>
<br>
<br>
<br>
<br>
<hr id="stopSpelling">Date: Tue, 1 Jun 2010 19:24:41 -0400<br>
From: <a class="moz-txt-link-abbreviated" href="mailto:zishanov@gmail.com">zishanov@gmail.com</a><br>
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] no sound between extensions<br>
<br>
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>
<br>
Zeeshan A Zakaria<br>
--<br>
Sent from my Android phone with K-9 Mail.<br>
<br>
<blockquote>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>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><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><br>
</div>
<br>
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