[Freeswitch-users] bridge a call to remote conference

[Madovsky]

I did more tests,
I can say that if the user bridged to the conference
has the same HZ rate of the conference (even if the codec is different)
so the sound is broken. 
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, January 16, 2011 2:33 PM
  Subject: Re: [Freeswitch-users] bridge a call to remote conference


  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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