[asterisk-ha-clustering] Active/Standby call interruption durring failover

[Marcin J. Kowalczyk]

Leif Madsen pisze:
> If you don't want to lose the audio, why not just re-invite the media
> from Asterisk? You'd essentially be doing the same thing without
> adding the complexity of OpenSER.
>   
 From http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

"Note: Asterisk is /still/ really not a SIP proxy in this case. The two 
legs have different Call-Ids, and so are different SIP calls. However 
the SDP descriptors for the audio of the two calls point directly at 
each other.

This short-circuit way of setting up calls was introduced in 1.4; for 
earlier versions, I believe the call is set up initially as two legs, 
and then Asterisk reconnects the endpoints directly using a re-INVITE "

I will retest on 1.6.x becouse on 1.4.x durring switch-over all cals 
were lost.
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