Hi Steve, thanks for your response...<br><br>I will try it this saturday and I'll let you know...<br><br>Best regards<br><br><div class="gmail_quote">On Wed, Jun 11, 2008 at 7:11 AM, Steve Totaro <<a href="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. <<a href="mailto:nachogomez@gmail.com">nachogomez@gmail.com</a>> wrote:<br>
> Hi list,<br>
><br>
> I'm having trouble with calls placed to the PSTN (through a TDM card),<br>
> sometimes (a lot indeed) when I dial a number the callee party can't hear me<br>
> at all.<br>
><br>
> My setup is:<br>
><br>
> Asterisk <a href="http://1.4.20.1" target="_blank">1.4.20.1</a><br>
> Zaptel 1.4.11<br>
> libpri 1.4.4<br>
> Wanpipe 3.2.4<br>
><br>
> I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP<br>
> Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel<br>
> 2.4.16.60-0.23-smp<br>
><br>
> I'm using the ulaw audio codec.<br>
><br>
> There is no NAT between the Asterisk Server and the Phones (the phone and<br>
> the server are in the same network segment).<br>
><br>
> What can it be???<br>
><br>
> Thanks in advance for any help/comment...<br>
><br>
><br>
> --<br>
> Raul<br>
> Linux Counter #156439<br>
<br>
</div></div>Is your Asterisk box dual homed? Firewalled? Any output from the CLI<br>
with verbose turned on, that might help? Turn on SIP debugging as<br>
well.<br>
<br>
Thanks,<br>
Steve T<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Nacho<br>Linux Counter #156439