[SR-Users] kamailio dialplan

[Hugh Waite]

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?
>
>
>    
>
> thanks,
>
> On Nov 8, 2013, at 6:13 PM, Fred Posner <fred at palner.com> wrote:
>
>> 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 at 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 at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Hugh Waite
Principal Design Engineer
Crocodile RCS Ltd.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20131111/ce5cee69/attachment.html>