<div dir="ltr">Maybe you have a Codec issue?<br><br><div class="gmail_quote">On Tue, Oct 28, 2008 at 11:20 PM, Benny Amorsen <span dir="ltr"><<a href="mailto:benny%2Busenet@amorsen.dk">benny+usenet@amorsen.dk</a>></span> 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">Lincoln King-Cliby <<a href="mailto:lincoln@controlworks.com">lincoln@controlworks.com</a>> writes:<br>
<br>
> Periodically I'm seeing calls placed from the 7961s through anything<br>
> on the PBX that requires digit entry (the Auto Attendant, Voicemail,<br>
> etc.) 'randomly' drop; extension-to-extension calls<br>
> extension-to-PSTN, and PSTN-to-extension calls never have any issues<br>
> whatsoever. Nor have I been able to duplicate the issues hopping<br>
> around auto attendants on an inbound PSTN call.<br>
<br>
</div>I am not sure this is relevant in the 1.4.x versions, but here goes<br>
anyway:<br>
<br>
In Asterisk 1.2.x it could sometimes happen that Asterisk believed the<br>
path to a server was so good, that it would only allow 1 ms for<br>
answers to be received. It would do all its retransmissions in less<br>
than 200ms, and then it would complain about no reply to critical<br>
packet.<br>
<br>
Anyway, you can adjust the minimum timer with the configuration option<br>
t1min in sip.conf. I would recommend setting it to at least 100 (it is<br>
in ms) and perhaps 500 would help for you.<br>
<br>
It is also highly possible that your issue is completely different.<br>
<font color="#888888"><br>
<br>
/Benny<br>
</font><div><div></div><div class="Wj3C7c"><br>
<br>
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