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

[Marcin J. Kowalczyk]

Leif Madsen pisze:
>
> 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.
>   
I've tried to use directrtpsetup and rtp goes directly between sip-client and mediagw (I can see it when I start rtp debug on asterisk-agent), my setup is following

mg0 (1.6.0.5) <=> asterisk-node0 <=> asterisk-agent (agent's PC) <=> sipClient

all * boxes has following setup:

canreinvite=yes
directrtpsetup=yes

but unfortunately restarting asterisk-node0 causes call to be dropped. On asterisk-node0 I'm using dial(Local/number at mg0) to dialout.
I wonder if putting OpenSer/Opensips in front of asterisk-node0 will change anything.

Best Regards,
Marcin

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