Hi,
In this configuration file, authentication is done before routing to Freeswitch.
If you are getting 407 responses it is because the request does not have an Authorization header or the credentials are incorrect. The cfg file you have attached does not have a realm defined in the calls to proxy_authorize/proxy_challenge (lines 520-521 and 505-507). This needs to contain the domain of your system, specifically, the one stored in the subscriber database. (See http://www.kamailio.org/docs/modules/devel/modules/auth.html#auth.f.proxy_challenge)
Authentication is done for every request, so if the client receives a 407 response, it must resend the request with the right credentials. You can check this is happening by looking at a tcpdump trace in wireshark or with ngrep.

The cfg file has some preconfigured routes for Freeswitch. If the called number does not match any of the regular expressions the call will probably be rejected with a 404 Not Found.
'43' does not match any of the expressions, but '433nnn' will forward to a FS conference.

Regards,
Hugh

On 11/11/2013 14:34, Joli Martinez wrote:

Hello, I log into the FS cli and set the cli to debug. I see not calls coming into FS when I dial 43. When I dial 41 I see the call hits the call and it get routed VM. I am new to Kamailio and don't know enough to start troubleshooting. If you can point me in the right direction. From what I read kamailio only has on config file and there is no reference to "43" in there so where do you configure the dial-peers or call routing?

When you dial 43 you get a prompt or 41? Also, do you see anything in the freeswitch logs or have a sip capture/ Fred Posner | The Palner Group direct: 503-914-0999 | fax: 954-472-2896 On 11/08/2013 06:04 PM, Joli Martinez wrote:

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. http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms#dokuwiki__top Also, since I am new could you give some pointers as far as security and documentation. thanks, _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users