(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:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; }
Currently with the above code if a user phones one of the other extensions it tries to route out to the PSTN network rather than the extension, is that because I have put the above code in the wrong place in the config so it never gets to the code to route to the extension? (routing to PSTN is fine)
Or do I need an if else statement wrap checking if local user, please can you give me some idea of the code ...
Thanks
Tony
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 20 May 2013 16:19 To: tony.turner@nodemax.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + Siremis Outbound route
Hello,
if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch:
if($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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Tony Turner