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velu.technical at gmai...
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 Posted: Mon Jul 27, 2009 12:28 am Post subject: [Freeswitch-users] IAX configurations |
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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/$1"/>
</condition>
</extension>
* Next I have configured a 222 user in sip.conf file at Asterisk machine.
* I wrote dialplan for that extension in extension.conf file.
When I tried to call 222 from FreeSWITCH, I have received following error in Console.
"[ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax"
What would be the problem? Is there any configuration I missed? Please help me .....
Regards,
K.Velusamy. |
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mike at jerris.com
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| Posted: Mon Jul 27, 2009 12:47 am Post subject: [Freeswitch-users] IAX configurations |
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mod_iax isn't loaded. I suggest using sip anyways.
Mike
On Jul 27, 2009, at 1:23 AM, velusamy velu wrote:
| Quote: | Dear All,
I have tried to call a Asterisk extension from FreeSWITCH. I have done the following configurations,
* I have enabled mod_iax module in modules.conf.xml file.
* Next I have configure following extension in dialplan.
<extension name="voipjet">
<condition field="destination_number" expression="^(222)$">
<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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velu.technical at gmai...
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| Posted: Mon Jul 27, 2009 1:08 am Post subject: [Freeswitch-users] IAX configurations |
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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 (mike@jerris.com)> wrote:
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william.suffill at gma...
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