[SR-Users] Trouble with Asterisk behind Kamailio

[Daniel Tryba]

On Wed, Jul 20, 2016 at 04:37:42PM -0400, Tickling Contest wrote:
> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
..
> but having trouble getting any further to load balance a couple
> of Asterisk servers. Here is a range of issues I have:
> 
> (a) The Kamailio server keeps sending UDP SIP messages to the Asterisk
> server and it is not clear where to control what protocol to use to send
> SIP messages Asterisk server(s).

route[TOASTERISK] creates $du in the following way:

$du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
			+ $sel(cfg_get.asterisk.bindport);

No mention of transport so it will use the default UDP.
Same for the other places where bindip/binport are used. If you want TCP
or TLS instead you'll have to add a transport to the URI or use
specific function to relay/send messages (like t_relay_to_tcp() instead
of t_relay()).

> (b) I followed the instructions for the dispatcher module in the wiki, but
> it is not clear why I should use the #!define WITH_ASTERISK directive to
> enumerate the bind IPs and ports (how does this reconcile with the list of
> Asterisk servers in the dispatcher.list file?):

You shouldn't since it doesn't reconcile. The WITH_ASTERISK directive is
used to communicate with 1 specific SIP server.
http://www.kamailio.org/docs/modules/stable/modules/dispatcher.html
contains a full example of a config with dispatcher. But note that the 
kamailio/asterisk realtime integration does some stuff (sending
registers to asterisk) that need to be handled differently (I suggest
useing kamailio as registar).