[Freeswitch-users] api conference dial

[Madovsky]

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 at 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 at 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 at domain.ltd 5555555555 freeSwitch

    to match default dialplan but the result is a loop.
    the log shows that the destination number is 9999999999 at domain.ltd-b,
    why a "-b" is added ? must I change my default dialplan to match it ?

    Thanks
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