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

[Guillaume Yziquel]

Steve Totaro a écrit :
> 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.

It was already 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.

Putting exten => _X.,n,Answer() in the dialplan doesn't change anything.

> That solution worked for me, but I could see how it could mess up CDRs and
> billing for some applications.

Maybe I'm having a different issue than you've been experiencing. What's 
rather painful is that nothing appears to show in the Asterisk CLI when 
this happens since it's obviously not a problem with the SIP connection.

How could I monitor the voice going in and out?

All the best,

Guillaume Yziquel.