[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)

[Vieri]

--- "Nestor A. Diaz" <nestor at tiendalinux.com> wrote:

> ok, thanks, does rtp*timeout work if i have
> canreinvite=yes ? since rtp 
> traffic is not passing thought asterisk, or i have
> to put canreinvite=no ?

In my setup it doesn't really matter since calls are
coming in through PSTN->IVR->QUEUE->SIP
AGENT->TRANSFERS THROUGH ZAP PRI TO ANOTHER PBX.



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