#!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
<mailto: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 <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
---
This email is free from viruses and malware because avast! Antivirus protection
is active.
http://www.avast.com
_______________________________________________
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
