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

28 May 2013

      Hello,
On 5/27/13 1:03 AM, Tony Turner wrote:
...
(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:" + $rU + "@__FREESWITCHIP__";
    route(RELAY);
    exit;
}
Siremis is a web interface for the database use for kamailio, not for 
kamailio configuration file. For using LCR or carrierroute, just read 
the documentation of these modules and add to your configuration file. 
LCR should be rather simple to setup, besides loading it and setting 
module parameters, you have to use two new functions in the routing 
logic of config file. Carrier route should be pretty much the same, just 
that I haven't checked it lately.
...
Currently with the above code if a user phones one of the other 
extensions it tries to route out to the PSTN network rather than the 
extension, is that because I have put the above code in the wrong 
place in the config so it never gets to the code to route to the 
extension? (routing to PSTN is fine)
Or do I need an if else statement wrap checking if local user, please 
can you give me some idea of the code  ...
If you don't have numbering rules that you can sort out if it is local 
subscriber or not, the best is to check with subscriber table (see 
uri_db or sqlops modules, both of them can help for this).
Cheers,
Daniel
...
Thanks
Tony
*From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com]
*Sent:* 20 May 2013 16:19
*To:* tony.turner@nodemax.com mailto:tony.turner@nodemax.com; 
Kamailio (SER) - Users Mailing List
*Subject:* Re: [SR-Users] Kamailio + Siremis Outbound route
Hello,
if you want to send all calls that arrive to kamailio having the 
prefix 01 to freeswitch:
if($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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-- 
Daniel-Constantin Mierla -http://www.asipto.com
http://twitter.com/#!/miconda  http://twitter.com/#%21/miconda  -http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
   *http://asipto.com/u/katu  *

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 - 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 *