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<br>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: zishanov@gmail.com<br>To: asterisk-users@lists.digium.com<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>
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<BR><blockquote>On 2010-06-01 7:02 PM, "Gary Baribault" <<a href="mailto:gary@baribault.net">gary@baribault.net</a>> wrote:<br><br>
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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>
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