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:(.*)$">
<action application="bridge"
data="{absolute_codec_string=c...@48000h@10i}sofia/${use_profile}/sip:
$1"/>
</condition>
</extension>
save that
then
pa call sip:[email protected]: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 <[email protected]>
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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