Sure Shaun, I'll give it a shot. I'll contact you directly to let you know the results.<br><br><div class="gmail_quote">On Wed, Feb 27, 2008 at 10:33 AM, Shaun Ruffell <<a href="mailto:sruffell@digium.com">sruffell@digium.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 class="Ih2E3d">arkda wrote:<br>
> Nothing in the console aside from what I've posted. When a DTMF tone is<br>
> played the server freezes instantly, hard reboot required.<br>
><br>
> Distribution is SuSE 10.2, kernel 2.6.18.8-0.7-default<br>
><br>
> The actual dialplan on this server is very simple, only one phone and a<br>
> few Dial commands via SIP to another server.<br>
><br>
> I've disabled as much as possible (including an extra NIC which botched<br>
> by G729 codecs) to try to eliminate any IRQ issues with no luck. I've<br>
> tried using various codecs including ulaw with dtmfmode=inband, but each<br>
> time the server freezes.<br>
<br>
</div>Would you be willing to try another branch of zaptel that is currently<br>
in beta? I would be interested if this version results in a server lock<br>
up with your configuration.<br>
<br>
The beta version is at:<br>
<a href="http://svn.digium.com/svn/zaptel/team/sruffell/voicebus" target="_blank">http://svn.digium.com/svn/zaptel/team/sruffell/voicebus</a><br>
<br>
If you want / need any help with this beta version, please contact me<br>
directly.<br>
<div><div></div><div class="Wj3C7c"><br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br>