i did as your said and got some errors in the console....
after i run pa call sip:886@ww2.bwrl.org:5080 i get.....
2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing relaxxplayer->sip:886@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@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@48000h@10i}sofia/${use_profile}/sip:$1"/>                                                
      </condition>                                                                                                                                              
    </extension>

save that
then


/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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