[asterisk-users] Outbound calls drop after 15 to 30 seconds.

[Steve Totaro]

On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel <
guillaume.yziquel at 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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