[asterisk-users] Weird one way Audio situation

[Steve Totaro]

On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. <nachogomez at gmail.com> wrote:
> Hi list,
>
> I'm having trouble with calls placed to the PSTN (through a TDM card),
> sometimes (a lot indeed) when I dial a number the callee party can't hear me
> at all.
>
> My setup is:
>
> Asterisk 1.4.20.1
> Zaptel 1.4.11
> libpri 1.4.4
> Wanpipe 3.2.4
>
> I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP
> Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel
> 2.4.16.60-0.23-smp
>
> I'm using the ulaw audio codec.
>
> There is no NAT between the Asterisk Server and the Phones (the phone and
> the server are in the same network segment).
>
> What can it be???
>
> Thanks in advance for any help/comment...
>
>
> --
> Raul
> Linux Counter #156439

Is your Asterisk box dual homed?  Firewalled?  Any output from the CLI
with verbose turned on, that might help?  Turn on SIP debugging as
well.

Thanks,
Steve T