<br><br><div class="gmail_quote">On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel <span dir="ltr"><<a href="mailto:guillaume.yziquel@citycable.ch">guillaume.yziquel@citycable.ch</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;">
Hello.<br>
<br>
I've set up and configured an Asterisk server to make SIP phone calls to<br>
external classic phones.<br>
<br>
However, it happens that after 15 or 30 seconds, the phone call drops.<br>
The SIP session still seems valid, but no sound comes through any more.<br>
<br>
How would you go through to troubleshoot this issue?<br>
<br>
All the best,<br>
<br>
Guillaume Yziquel.<br>
<br>
</blockquote><div><br>Make sure you have canreinvite set to no.<br><br>Also, you may need to put an answer() in before your dial, I have dealt with that strangeness, call always drop at exactly 30 seconds. <br><br>That solution worked for me, but I could see how it could mess up CDRs and billing for some applications.<br>
</div></div><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br>