--- "Nestor A. Diaz" <[EMAIL PROTECTED]> 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
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 ?
slds.
> rtp*timeout for sip peers is not a fix but a
> workaround.
> Try to set both values and reload sip.
> Then when you witness what you posted tr
--- "Nestor A. Diaz" <[EMAIL PROTECTED]> wrote:
> Vieri wrote:
> > Did you try a "show channels" to see if there were
> > stale channels for peer 200?
> >
> > I had the same problem you describe but it was due
> to
> > "hung" channels (used * 1.4.18.1 with rtp*timeout
> and
> > saw "inuse" peers
Vieri wrote:
> Did you try a "show channels" to see if there were
> stale channels for peer 200?
>
> I had the same problem you describe but it was due to
> "hung" channels (used * 1.4.18.1 with rtp*timeout and
> saw "inuse" peers during the pre-timeout periods even
> though the agents weren't on a
--- "Nestor A. Diaz" <[EMAIL PROTECTED]> wrote:
> Mojo with Horan & Company, LLC wrote:
> > Nestor A. Diaz wrote:
> >
> >> 1. I use a queue with just on sip device, one
> call at a time, however
> >> and without reason just after some couple of
> hours the sip device show
> >> in use and the
Mojo with Horan & Company, LLC wrote:
> Nestor A. Diaz wrote:
>
>> 1. I use a queue with just on sip device, one call at a time, however
>> and without reason just after some couple of hours the sip device show
>> in use and then no calls are transfered from the queue to the sip
>> device, i
Nestor A. Diaz wrote:
> 1. I use a queue with just on sip device, one call at a time, however
> and without reason just after some couple of hours the sip device show
> in use and then no calls are transfered from the queue to the sip
> device, i do a sip show inuse and this is the result:asteri
Hello Asterisk People,
I have two annoying bugs in asterisk, that i want to know if some of you
have already found a way to fix:
Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch.
1. I use a queue with just on sip device, one call at a time, however
and without reason jus
