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Leif Madsen pisze:<br>
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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.
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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@mg0) to dialout.
I wonder if putting OpenSer/Opensips in front of asterisk-node0 will change anything.
Best Regards,
Marcin
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