[Freeswitch-users] how to use celt codec

[e schmidbauer]

i did as your said and got some errors in the console....
after i run pa call sip:886 at ww2.bwrl.org:5080 i get.....
2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
relaxxplayer->sip:886 at ww2.bwrl.org:5080 in context default
2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid
Profile
2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[DESTINATION_OUT_OF_ORDER]
2008-12-30 14:45:25 [INFO] mod_dptools.c:1891 audio_bridge_function()
Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER

any suggestions?


On Tue, Dec 30, 2008 at 2:06 PM, Brian West <brian at freeswitch.org> wrote:

> 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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