t_on_failure("F_VOIP") to be used before t_relay();
That will arm the call to go to F_VOIP on failure responses.
On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran <alijib...@vividtech.io> wrote:
>
>
> #!ifdef WITH_FREESWITCH
>
> if(is_method("INVITE") && route(FROMFREESWITCH))) {
>
> xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call from
> FreeSWITCH needs to be sent TOVOIP \n");
>
> route(TOVOIP);
>
> t_on_failure("F_VOIP");
>
> exit;
>
> }
>
>
>
> #!endif
>
>
>
>
>
>
>
> route[TOVOIP] {
>
> xlog("L_INFO","ALERT: $fu to $tu ");
>
> $fU="XXXXXX";
>
> $td="sip.voipfone.net";
>
> $du="sip:xxxx...@sip.voipfone.net";
>
> t_relay();
>
>
>
> }
>
>
>
>
>
> failure_route[F_VOIP] {
>
> uac_auth();
>
> xlog("L_INFO","ALERT: IN FAIL");
>
> }
>
>
>
>
>
> I tried this but it never makes it to the failure branch. Im a newbie to
> kamailio and still working around the scripting. Can you please help me out
> here to where I am making the mistake?
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *SamyGo
> *Sent:* Thursday, April 30, 2015 9:18 AM
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] UAC Module
>
>
>
> 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
> <http://opensips.org/pipermail/users/2010-August/013947.html>):
>
>
>
> 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
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