[asterisk-users] SIP call control via RTCP

[Jan Gaida]

Hello,

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?

Kind regards


-- 
    *Jan **Gaida*
Ingeniero Desarrollo Software     C/ Marconi 3 (PTM)
28760 Tres Cantos
Spain
jan.gaida at grupoamper.com | www.grupoamper.com

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