Thanks, any reason I could not just use FreeSwitch as an SBC and then load calls between multiple Asterisk servers? Thanks, Keith

On Wed, Feb 20, 2013 at 7:12 PM, Andreas Granig agranig@sipwise.com wrote:

Hi,

On 02/20/2013 04:24 PM, Radek Wal wrote:

...

SEMS als SBC application.

You might want to check http://www.sipwise.com/news/** technical/byov-system-spce-as-**sbc/http://www.sipwise.com/news/technical/byov-system-spce-as-sbc/which uses kamailio and sems to do exactly that.

Andreas

So in my setup where FreeSwitch and Kamailio sit on the same physical server:

Kamailio - Routes traffic to asterisk boxes Freeswitch - SBC and RTP traffic B2BUA

Question is, can I load balance my voice traffic to the asterisk servers?

From what I have read you can only do this with a percentage rather than

based on the amount of traffic currently on each server (true load balancing).

Thanks, Keith

On Tue, Feb 26, 2013 at 8:41 AM, Alex Balashov abalashov@evaristesys.comwrote:

There is no reason, other than the "soft" question of FreeSwitch's suitability to the SBC role, as the term has been defined by the industry, and the inertial forces of custom and habit.

Keith keithhubner@gmail.com wrote:

...

Thanks, any reason I could not just use FreeSwitch as an SBC and then load calls between multiple Asterisk servers? Thanks, Keith

On Wed, Feb 20, 2013 at 7:12 PM, Andreas Granig agranig@sipwise.comwrote:

...

Hi,

On 02/20/2013 04:24 PM, Radek Wal wrote:

...

SEMS als SBC application.

You might want to check http://www.sipwise.com/news/** technical/byov-system-spce-as-**sbc/http://www.sipwise.com/news/technical/byov-system-spce-as-sbc/which uses kamailio and sems to do exactly that.

Andreas

-- Keith

---

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

-- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

Hello

Or start with this tutorial:

http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb

later You can change it to use dispatcher module to have load balance between couple od Asterisk GW.

But try to change configuration to use WITH_PSTN, not WITH_ASTERISK, that won't forward registration to Asterisk, only INVITE's, in PSTN Route implement dispatcher module.

Greetings Andrzej

Cytowanie davy van de moere davy.van.de.moere@gmail.com:

Stampeding on an open door here, and not wanting to start a pun war ;)

http://www.amazon.com/Building-Telephony-Systems-OpenSIPS-1-6/dp/1849510741/...

Offcourse, it's not *AS* good as Kamailio, but if your into the book thing, it might help you to get your head around how the thing actually works.

The question you ask, is actually a close to default setting of Kamailio, so I think you need to or:

1/ get your head around routing sip packets, which at starters is mindblowing, so give yourself some time. 2/ pay someone to set a kamailio up for you which does what you want. (I think almost anyone on the mailinglist here, can do that in a matter of hours)

Enjoy!

On 19 Feb 2013, at 10:52, Keith wrote:

...

Hi, I am looking to build an SBC/SIP router made up from Kamailio and FreePBX. Calls will then need to be passed to multiple Asterisk media gateways. Can anyone point me to some documentation or help on how to do this?

-- Keith _______________________________________________ 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