Hello,
if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch:
if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; }
Be sure calls are authenticated at that point and, if needed, the call is not actually coming from freeswitch.
Cheers, Daniel
On 5/20/13 11:33 AM, Tony Turner wrote:
Hi
Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get install
I want to use Kamailio as a proxy edge register to our network.
I have installed Kamailio and freeswitch.
I can register on Kamailio but I can't route a call from my sip client from Kamailio to freeswitch and out to PSTN
Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- Carriers
If I register direct on Freeswitch I can route out to PSTN but I don't understand Kamailio routing.
Can someone let me how I route say from SIP client registered on Kamailio to prefix 01% which goes out to Freeswitch
Many Thanks
Tony
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
I have 3 registered test users, how can I set up Siremis to do the trunk to freeswitch rather than using LCR or Carrierroute, are there any Siremis instructions for either ...
if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; }
Currently with the above code if a user phones one of the other extensions it tries to route out to the PSTN network rather than the an extension is that because I have put the above code in the wrong place in the config.
Or do I need an if else statement checking if local user, can you give me some idea of the code ...
Thanks
Tony
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 20 May 2013 16:19 To: tony.turner@nodemax.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + Siremis Outbound route
Hello,
if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch:
if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; }
Be sure calls are authenticated at that point and, if needed, the call is not actually coming from freeswitch.
Cheers, Daniel
On 5/20/13 11:33 AM, Tony Turner wrote:
Hi
Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get install
I want to use Kamailio as a proxy edge register to our network.
I have installed Kamailio and freeswitch.
I can register on Kamailio but I can't route a call from my sip client from Kamailio to freeswitch and out to PSTN
Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- Carriers
If I register direct on Freeswitch I can route out to PSTN but I don't understand Kamailio routing.
Can someone let me how I route say from SIP client registered on Kamailio to prefix 01% which goes out to Freeswitch
Many Thanks
Tony
_______________________________________________ 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
-
Daniel-Constantin Mierla -
Tony Turner