I have loaded mod_iax now that error didn't come. But, When I call I have received following message in the console. "[INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: FACILITY_REJECTED" What configuration I missed? How to use sip to connect the Asterisk? Please give solutions above questions... __ Velusamy On Mon, Jul 27, 2009 at 11:11 AM, Michael Jerris <mike@jerris.com> wrote:
mod_iax isn't loaded. I suggest using sip anyways. Mike
On Jul 27, 2009, at 1:23 AM, velusamy velu wrote:
Dear All, I have tried to call a Asterisk extension from FreeSWITCH. I have done the following configurations, * I have enabled mod_iax module in modules.conf.xml file. * Next I have configure following extension in dialplan. <extension name="voipjet"> <condition field="destination_number" expression="^(222)$"> <action application="bridge" data="iax/222:222@192.168.6.94/$1"/> </condition> </extension> * Next I have configured a 222 user in sip.conf file at Asterisk machine. * I wrote dialplan for that extension in extension.conf file.
When I tried to call 222 from FreeSWITCH, I have received following error in Console. "[ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax"
What would be the problem? Is there any configuration I missed? Please help me .....
Regards, K.Velusamy.
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