[asterisk-users] Timeout when dialing dead peer

[Klaus Darilion]

Benny Amorsen schrieb:
> Stefan Schmidt <sst at sil.at> writes:
> 
>> if i understand you right you have one server (peer) where thousands of
>> devices are connected and every device is registered to asterisk, and so
>> every options packet will come from asterisk to this device, right?
>> If you have a sip routing server like ser, the server itself could do a
>> Nat keep alive check, and could drops the invite coming from asterisk if
>> the peer isnt reachable. If these devices arent registered to asterisk
>> why do you think that there will be so much options Packets? if you have
>> one peer this will get only one Options packet per minute.
> 
> I have decided to go with the qualify solution as you recommend. I have
> set the qualify frequency to 1800 seconds; that way I should get the
> lowered t1 timing with very few qualify requests.
> 
> It looks like it works, thanks!

You can also set the overall timeout with timerb:

;timerb=32000   ; Call setup timer. If a provisional response is not 
received
                 ; in this amount of time, the call will autocongest
                 ; Defaults to 64*timert1




regards
klaus