Hi list,
I'm new here, also to Kamailio. My goal is to achieve the "standard
solution": Kamailio for load balancing and high availability of several
Asterisk servers.
For starters, I've set it up with Kamailio+dispatcher+permissions acting
as a proxy and it was working fine, but I'd like to try to keep it much
more simple. For these reasons:
1) I'd like to maintain different peer settings in Asterisk (DTMF, codecs).
2) I don't want/need a database for Kamailio. I think it is mandatory
for permissions module + IP auth. So, let's do Asterisk the
authentication, it's already there.
3) Accounting. I'm using a software where it's really useful for
accounting purposes if every peer/user has its own entry in Asterisk's
sip.conf / in realtime table.
Long story short, my config now looks like this and it seems to be
working fine for me:
request_route {
route(DISPATCHER);
}
route[DISPATCHER] {
if ( method=="REGISTER") {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
if ( method=="INVITE" ) {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
if ( method=="SUBSCRIBE" ) {
ds_select_domain("1","4");
sl_send_reply("300","Redirect");
exit;
}
}
My questions:
1) There are other approaches to the DTMF+codecs issue. Like this thread
from 2013:
http://lists.kamailio.org/pipermail/sr-users/2013-January/076379.html -
but I have dozens of changing PSTN peers, so authentication/accounting
based on port would be cumbersome. Are there new/alternative ways?
2) I searched the archive and the Web but did not find many questions
about a "I-don't-want-Kamailio-to-show-as-the-only-peer-to-Asterisk"
solution. Therefore I'm wondering: Is everyone a) running such
homogeneous setups that they can survive with the same codec + DTMF
settings in Asterisk for all their peers + users? Or b) doing some
rewriting magic within Kamailio to deal with it? Or c) I'm overlooking a
simple solution. Which one is it? :)
3) And the biggest question: Everywhere I read basically "Redirect
server for LB + HA is not recommended, better use proxy". Is this still
true, and if so, what are the reasons? Because I should be worried that
not all clients/gateways out there respect a SIP 300 Redirect? Other
reasons?
Thanks so much! :)
Markus
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