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

[Leif Madsen]

On Wed, Feb 4, 2009 at 9:55 AM, Marcin J. Kowalczyk
<marcin.kowalczyk at ccig.pl> wrote:
> 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.

Try using 'directrtpsetup' which doesn't send the initial audio
through the Asterisk systems. Note this is still considered
experimental, but I haven't seen any bugs reported with it, and I've
had good success with it in the simple scenarios I've given it to.

Leif Madsen.