[SR-Users] Kamailio + Siremis Outbound route ---- Re Sent Due to bad English

[Tony Turner]

(re-sent due to bad English)

 

Hi,

 

I have 3 registered test users, how can I configure Siremis to do the trunk
to freeswitch using LCR or Carrierroute rather than using the code below. I
am keen to be able to setup Inbound + Outbond trunks via Siremis. Do you
know if there is a manual for Siremis or a how to / step by step?

 

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 extension, is that
because I have put the above code in the wrong place in the config so it
never gets to the code to route to the extension? (routing to PSTN is fine)

 

Or do I need an if else statement wrap checking if local user, please can
you give me some idea of the code  ...

 

Thanks

Tony 

 

 

 

From: Daniel-Constantin Mierla [mailto:miconda at gmail.com] 
Sent: 20 May 2013 16:19
To: tony.turner at 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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-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
  * http://asipto.com/u/katu *
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