[Freeswitch-users] how to use celt codec

Brian West brian at freeswitch.org
Tue Dec 30 11:06:10 PST 2008


OK here try this..

in portaudio.conf.xml

     <param name="sample-rate" value="48000"/>
     <param name="codec-ms" value="10"/>


in dialplan/default.xml

     <extension name="sip_uri">
       <condition field="destination_number" expression="^sip:(.*)$">

         <action application="bridge"  
data="{absolute_codec_string=CELT at 48000h@10i}sofia/${use_profile}/sip: 
$1"/>
       </condition>
     </extension>

save that
then

pa call sip:886 at taz.bkw.org:5080

/b


On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:

> i have port audio setup but when i do a 'pa call <extension>' it  
> enters the conference using the L16 codec. is there a way to use  
> celt codec instead of the L16?
>
> On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian at freeswitch.org>  
> wrote:
> http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> /b
>
> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
>
> > Could you explain in a more detail how you set that up?
>
>
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