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 monitor the
RTP stream and then initiate a BYE if it sees no RTP packets coming in.
That way Asterisk will receive the proper disconnect signal in a
canreinvite=yes scenario.
David
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 idea is this.. I want to use
canreinvite =yes to force users to talk end-to-end to preserve
bandwidth, but I can see the potential for hung calls if asterisk
never get the BYE from a UA in the event the ATA gets unplugged or
somehow loses power.
That is the case in every SIP network, Asterisk is not unique in that
regard.
I would suggest that you could make a modification to chan_sip so that
if the peer goes 'unreachable' (as determined by using qualify=yes)
than any existing calls involved with that peer are immediately hung
up; that would take care of this problem.
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