[SR-Users] kamailio dialplan

Joli Martinez mrjoli021 at gmail.com
Sat Nov 9 00:04:51 CET 2013

I am new to Kamailio and am having an issue with the dialplan setup.  I have Kamailio setup as an SBC to handle all user authentication and call routing.  I need freeswitch to handle all conferences and voicemails.  When I dial 433001 I would like to be transferred to freeswitch for conferences.  Right now I have followed the following article and it when I dial 433001 call hangs up and never reaches FS.  If I call 43 call does reach FS and I am able to hear FS play the VM prompt.  

My system is CentoOS 6.4 and FS is installed via yum, but Kamailio is complied. Both FS and Kamailio are on the same box.

What commands would you suggest I use to troubleshoot these issues in the future.


Also, since I am new could you give some pointers as far as security and documentation.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20131108/0aefcc83/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: kamailio.cfg
Type: application/octet-stream
Size: 16417 bytes
Desc: not available
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20131108/0aefcc83/attachment.obj>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20131108/0aefcc83/attachment-0001.html>

More information about the sr-users mailing list