[asterisk-dev] rtp timeout failover

[Olle E. Johansson]

1 jun 2010 kl. 15.45 skrev Wolfgang Pichler:
> 
> call leg A does dial call leg B over a channel with rtp data. call leg
> B rtp data does timeout -> so call leg A continues at
> context/extension/prio...
> 

> 
> I am currently locking at chan_sip from asterisk 1.4.21 - there around
> line 15725 is the rtptimeout handling - as far as i can see it does
> simple hang up the call leg where the rtp timeout did occoured - so
> dialplan execution will continue for call leg A - but it will look
> like the hangup was a normal hangup. What about adding a new channel
> variable to call leg A - which does mean that the hangup is because of
> a rtp timeout ?
The channel variable would then be set in the outbound channel that we
will destroy...
> 
> An other possibility would be to create a new HANGUP Cause for this -
> but i think this change would have many side effects... - so not the
> best way
THat may be the only useful way. What cause are you getting today?
Anything useful?

/O