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:(.*)$"> <action application="bridge" data="{absolute_codec_string=CELT@48000h@10i}sofia/${use_profile}/sip: $1"/> </condition> </extension> save that then pa call sip:886@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@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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