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

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