I have loaded mod_iax now that error didn't come.<br>But, When I call I have received following message in the console.<br>"[INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: FACILITY_REJECTED"<br>
<br>What configuration I missed?<br>How to use sip to connect the Asterisk?<br><br>Please give solutions above questions...<br>__<br>Velusamy<br><br><br><div class="gmail_quote">On Mon, Jul 27, 2009 at 11:11 AM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</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="">mod_iax isn't loaded. I suggest using sip anyways.<div><br></div><div>Mike</div>
<div><br><div><div><div></div><div class="h5"><div>On Jul 27, 2009, at 1:23 AM, velusamy velu wrote:</div><br></div></div><blockquote type="cite"><div><div></div><div class="h5">Dear All,<br> I have tried to call a Asterisk extension from FreeSWITCH. I have done the following configurations,<br>
* I have enabled mod_iax module in modules.conf.xml file.<br> * Next I have configure following extension in dialplan.<br> <extension name="voipjet"><br>
<condition field="destination_number" expression="^(222)$"><br> <action application="bridge" data="iax/<a href="http://222:222@192.168.6.94/$1" target="_blank">222:222@192.168.6.94/$1</a>"/><br>
</condition><br> </extension><br> * Next I have configured a 222 user in sip.conf file at Asterisk machine.<br>
* I wrote dialplan for that extension in extension.conf file.<br> <br> When I tried to call 222 from FreeSWITCH, I have received following error in Console. <br> "[ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax"<br>
<br> What would be the problem? Is there any configuration I missed? Please help me .....<br><br>Regards,<br>K.Velusamy.<br><br><br></div></div> _______________________________________________<br>FreeSWITCH-users mailing list<br>
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