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 <m...@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/222:2...@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. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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