[asterisk-users] Timeout when dialing dead peer

[Benny Amorsen]

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!


/Benny