[asterisk-users] SIP call control via RTCP

[Matt Behrens]

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?  

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