So after some invite tests I notced 6/8 seconds of audio
delay between the invited call and conference.
if the invited leg call himself the conference the latency is reasonable
I heard in previous threads that there was maybe a latency problem
if loopback is used in conference ?
 
----- Original Message -----
From: Madovsky
To: freeswitch-users@lists.freeswitch.org
Sent: Sunday, March 27, 2011 1:07 PM
Subject: Re: api conference dial

sorry forget my request.
I needed to set sip_to_uri variables to match my default dialplan correctly
 
Thanks
----- Original Message -----
From: Madovsky
To: freeswitch-users@lists.freeswitch.org
Sent: Sunday, March 27, 2011 12:26 PM
Subject: api conference dial

I try to do this
 
conference testconf dial loopback/9999999999@domain.ltd 5555555555 freeSwitch
 
to match default dialplan but the result is a loop.
the log shows that the destination number is 9999999999@domain.ltd-b,
why a "-b" is added ? must I change my default dialplan to match it ?
 
Thanks