i did as your said and got some errors in the console....<br>after i run pa call <a>sip:886@ww2.bwrl.org:5080</a> i get.....<br>2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing relaxxplayer-><a href="http://sip:886@ww2.bwrl.org:5080">sip:886@ww2.bwrl.org:5080</a> in context default<br>
2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid Profile<br>2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW]<br>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]<br>
2008-12-30 14:45:25 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER<br><br>any suggestions?<br><br><br><div class="gmail_quote">On Tue, Dec 30, 2008 at 2:06 PM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div style="">OK here try this.. <div><br></div><div>in portaudio.conf.xml</div><div><br></div>
<div><div> <param name="sample-rate" value="48000"/> </div><div> <param name="codec-ms" value="10"/></div>
<div><br></div><div><br></div><div>in dialplan/default.xml</div><div><br></div><div><div> <extension name="sip_uri"> </div>
<div> <condition field="destination_number" expression="^<a>sip:(.*)$</a>"> </div><div> </div>
<div> <action application="bridge" data="{absolute_codec_string=CELT@48000h@10i}sofia/${use_profile}/<a>sip:$1</a>"/> </div><div> </condition> </div>
<div> </extension></div><div><br></div><div>save that</div><div>then</div><div><br></div><div>pa call <a>sip:886@taz.bkw.org:5080</a></div><div><br></div><div>/b</div><div><br></div></div></div><div><br><div><div class="Ih2E3d">
<div>On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:</div><br></div><div><div></div><div class="Wj3C7c"><blockquote type="cite">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?<br>
<br><div class="gmail_quote">On Tue, Dec 30, 2008 at 1:36 PM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<a href="http://wiki.freeswitch.org/wiki/Freeswitch_softphone" target="_blank">http://wiki.freeswitch.org/wiki/Freeswitch_softphone</a><br> <font color="#888888"><br> /b<br> </font><div><br> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:<br>
<br> > Could you explain in a more detail how you set that up?<br> <br> <br> </div><div><div></div><div>_______________________________________________<br> Freeswitch-users mailing list<br> <a href="mailto:Freeswitch-users@lists.freeswitch.org" target="_blank">Freeswitch-users@lists.freeswitch.org</a><br>
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