[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?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20081230/5e5a506a/attachment-0001.html
More information about the Freeswitch-users
mailing list