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