[SR-Users] ovh trunk
gled at remote-shell.net
Wed Feb 24 21:16:24 CET 2016
This won't answer your question, but I would rather use an sbc (
freeswitch, yate, asterisk, ...) to deal with the routing logic,
especially if you
want to leave some room for the future ( like transcoding, retrying a
call to a different carrier, ... ).
Kamailio can then be used as an awesome load-balancer/proxy in front of
your sbc farm.
That said, if you don't want/need to deal with the media, and just want
to work with the sig part, then you should look at the Uac module.
PS: The OPTION packets you saw are probably NAT keepalives that you can
On 02/24/2016 12:04 PM, Sébastien Brice wrote:
> Hi everyone
> I followed this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
> and got it working (101, 102 and 103) can call eachother.
> But now i am trying to figure Asterisk's role out.
> I am more an ipbx person and i am used to register providers trunk in asterisk/sip.conf file, like this:
> register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension]
> doing that i got plenty of OPTIONS request and 200 OK reply between my Kamailio and my provider (and it is a bit noisy)
> doing that i feel missing kamailio's logic and power to deal with externals trunk provider
> The thing is i need my authenticated users (101,102,103) be capable dialing my trunk and requesting INVITE for non-local request.
> What is the best way to achieve that?
> My DID provider gave me user/passwd/realm.
> I heard about avp special variables (auth_XXXX_avp and uac) and some snippets config that could help me to go there.
> Is that efficient to place the routing's logic to Kamailio and how to do that with my ovh trunk?
> thx you
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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