[Freeswitch-users] bridge a call to remote conference

[Madovsky]

more weird,

if I choose a codec from the sip phone based on 8000hz it doesn't work.
it's like a rate conversion has to be forced to make the audio work on the conference bridge...
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Sunday, January 16, 2011 12:10 PM
  Subject: Re: [Freeswitch-users] bridge a call to remote conference


  after some test I confirm that it should force to offer a codec
  at the same rate of the conference, example

  <action application="bridge" data="{absolute_codec_string=GSM,ignore_early_media=true}sofia/internal/${conference}"/>

  as GSM is a 8000hz code so it works now...


  thanks
    ----- Original Message ----- 
    From: Steven Ayre 
    To: FreeSWITCH Users Help 
    Sent: Sunday, January 16, 2011 4:20 AM
    Subject: Re: [Freeswitch-users] bridge a call to remote conference


    Try answering the call before you enter the conference, possibly with a small sleep inbetween too. That might give the media time to startup before the conference IVR starts.

    http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer
    http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep

    -Steve



    On 16 January 2011 03:54, Madovsky <infos at madovsky.org> wrote:

      everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec)
      but the hold music (in 8000hz) works well.
      should I set any special codec in the legB conference bridge ?

      Thanks

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