[asterisk-users] Calls drop after 20 seconds

[Stefan Schmidt]

Alejandro Recarey schrieb:
>> Stefan
>> How do you dial the users? direct with the peername or something like
>> exten at ipofpeer ?
>>
>> i know this problem when dialing a patton ISDN ata without an extension.
>> The call is established but when the T1 sip timeout fires the call gets
>> disconnected. Maybe you could do some sip debugging and watch for resend
>> sip messages.
>>     
>
> I don't understand, all of my calls are inbound and terminated with
> different voip carriers, so I am not sure how that will work. I always
> dial dst at ipofcarrier. Will debug!
>   
what i mean is that the problem what i have was when i dial no exten 
directly via the IP of the patton.

which looks like this: Dial(SIP/@123.123.123.123,120)
when this happens the T1 Timeout ends the call after 30 seconds.
this only happens on inbound calls to the customers, not outbound to a 
carrier.


best regards.

steve