Thanks for the information Matt!
Does asterisk store any SIP dialog cdr info in mysql like Call-ID &
Cseq? With This info I could at least detect runaway calls and fake a
BYE to the pstn gateway with an external app.
regards,
David
On 11/23/05, Matt Riddell <[EMAIL PROTECTED]> wrote:
> David Tho
Kevin,
Is the CDR accounting done based on SIP signaling? If a UA is talking
(RTP) to a third party PSTN gateway, isn't it at risk if say the UA
loses power. How will asterisk know the call has ended if it is not
involved in the media path. The idea is this.. I want to use
canreinvite =yes to forc
Does asterisk have support for SIP session timers?
David
On 11/24/05, Olle E. Johansson <[EMAIL PROTECTED]> wrote:
> Matt Riddell wrote:
> > Kevin P. Fleming wrote:
> >
> >>Matt Riddell wrote:
> >>
> >>
> >>>So how does Asterisk know that the media stream has been disconnected
> >>>between
> >>>t
David Thomas wrote:
Is the CDR accounting done based on SIP signaling? If a UA is talking
(RTP) to a third party PSTN gateway, isn't it at risk if say the UA
loses power. How will asterisk know the call has ended if it is not
involved in the media path. The idea is this.. I want to use
canreinvi
If you use qualify=yes to determine whether that device is alive or not,
then it won't be very accurate as every now and then, the device may
fail to reply to the SIP OPTIONS packet due to reasons other than it is
really offline.
If you are linked to a PSTN GW, I would believe that GW will mon
Kevin P. Fleming wrote:
> David Thomas wrote:
>
>> Is the CDR accounting done based on SIP signaling? If a UA is talking
>> (RTP) to a third party PSTN gateway, isn't it at risk if say the UA
>> loses power. How will asterisk know the call has ended if it is not
>> involved in the media path. The
David Thomas wrote:
> Does asterisk have support for SIP session timers?
>
No.
/O
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