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
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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