On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel < guillaume.yziq...@citycable.ch> wrote:
> Hello. > > I've set up and configured an Asterisk server to make SIP phone calls to > external classic phones. > > However, it happens that after 15 or 30 seconds, the phone call drops. > The SIP session still seems valid, but no sound comes through any more. > > How would you go through to troubleshoot this issue? > > All the best, > > Guillaume Yziquel. > > Make sure you have canreinvite set to no. 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. That solution worked for me, but I could see how it could mess up CDRs and billing for some applications. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)
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