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 <govoi...@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 <alijib...@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 >> _______________________________________________ >> 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
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