[asterisk-users] SIP call control via RTCP
matt at zigg.com
Mon May 12 07:12:02 CDT 2014
On May 12, 2014, at 5:02 AM, Jan Gaida <jan.gaida at grupoamper.com> wrote:
> We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls.
> With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no mechanism to check whether a call is still active.
> Is there a way to activate a RTCP call control, e.g. Asterisk should hang up when he stops receiving RTCP messages?
Have you looked at the rtptimeout and rtpholdtimeout options?
-------------- next part --------------
A non-text attachment was scrubbed...
Size: 801 bytes
Desc: Message signed with OpenPGP using GPGMail
More information about the asterisk-users