Copy your dialplan and sip debug a call.<div><br></div><div><br><br><div class="gmail_quote">On Sun, Nov 2, 2008 at 3:07 PM, Jim Boykin <span dir="ltr"><<a href="mailto:boykinjim@gmail.com">boykinjim@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Any help. Thanks<br>
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On Sun, Nov 2, 2008 at 12:50 PM, Jim Boykin <<a href="mailto:boykinjim@gmail.com">boykinjim@gmail.com</a>> wrote:<br>
> Marcin, can you elaborate. No timer has been set and call is not idle either.<br>
><br>
> Thanks<br>
> Jim<br>
><br>
> On Fri, Oct 31, 2008 at 5:17 PM, Marcin J. Kowalczyk<br>
> <marcin.kowalczyk@ccig.pl> wrote:<br>
>> Jim Boykin pisze:<br>
>>> We are running Asterisk SVN. We are facing a strange and repetable<br>
>>> problem. All outgoing call gets terminated in approx 20 minutes.<br>
>>> Asterisk initiates BYE message to the remote end and call terminates.<br>
>>><br>
>> Sesion-timer set but not supported by sip-peers?<br>
>><br>
>><br>
>><br>
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><br>
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