[SR-Users] UAC Module

[Ali Jibran]

Thanks for the awesome detailed explanation :)

I talked to Voipfone(trunk) and they only allow registered endpoints to make/receive calls. So I can't do IP Auth as of now. 

I'll try the other method by rewriting $fu and $du. Hopefully that'll work.

Thanks for the help again. 

AJ

> On 30-Apr-2015, at 9:18 am, SamyGo <govoiper at gmail.com> wrote:
> 
> Hi Jibran,
> 
> Here is an old thread as reference:
> 
> http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
> 
> I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication. 
> Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
> 
> From the configuration perspective this is my idea of still using UAC.
> 
> - Call coming from FS on kamailio
> - Rewrite the from-uri  (so the provider receives calls from the registered username)
> - modify the to-domain part to contain the IP address of the provider
> - set the $du to ip of the provider, and t_relay() the call.
> - Most likely the Provider would say Proxy-Auth required..that can be caught in failure_route[]
> - There you can call the uac_auth() function to have username.password attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
> - once this function is successful send the INVITE again to the provider.
> 
> Last three steps can be the following snippet of code(reference from here):
> 
> failure_route[2] {
>      if (t_check_status("40[17]")) {
> 	xlog("got challenged \n");
>      	if (uac_auth()) {
> 	    xlog("auth was succesful \n");
>             t_relay("udp:ip_addr:5060"); //provider's IP_ADDR
> 	}
> }
> 
> 
> I hope you get IP Auth from the provider, and find the reply useful.
> 
> Regards,
> 
> 
> 
>> On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran <alijibran at vividtech.io> wrote:
>> 
>> Hi all.
>> I have this setup.
>> Trunk--->Kamailio---->FreeSWITCH
>> 
>> I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
>> 
>> I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
>> 
>> Basically this is what I'm trying to workout
>> FS---->kamailio---->trunk.
>> 
>> 
>> Any help will be much appreciated. Thanks.
>> AJ
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> 
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