[Freeswitch-users] IAX configurations

[velusamy velu]

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 at 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 at 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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