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.mo...@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/ref=sr_1_1?ie=UTF8&qid=1361373796&sr=8-1&keywords=kamailio
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
_______________________________________________
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
