[asterisk-users] no sound between extensions

[Gary Baribault]

I don't know if this makes any difference, I created a lot of this
configuration with the Asterisk-GUI (SVN-branch-2.0-r4980) and when I
edit the users.conf file, there are two entries 'type = peer' for each
extension and they are highlighted in red!

Gary Baribault

On 06/02/2010 08:32 AM, taimur hasan wrote:
>
> 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).
>
> *-THQ-  !!!ONE*
>
>
>
>
>
> ------------------------------------------------------------------------
> Date: Tue, 1 Jun 2010 19:24:41 -0400
> From: zishanov at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] no sound between extensions
>
> 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'.
>
> Zeeshan A Zakaria
> --
> Sent from my Android phone with K-9 Mail.
>
>     On 2010-06-01 7:02 PM, "Gary Baribault" <gary at baribault.net
>     <mailto:gary at baribault.net>> wrote:
>
>     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.
>
>     Gary Baribault
>
>     On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
>     >
>     > Output of 'iptables -L -n' would also be helpfu...
>
>     --
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