Stoyan thanks for your reply, i've been doing some research before replying (which has taken a while) and there's something I don't understand. I apologize in advance if I'm asking something that makes no sense.
my provider does in fact requires registration, and they provide a single sip that accepts hundreds of concurrent calls.
If I have a kamailio in front of several pbx servers, in order to have redundancy (if server fails) and be able to handle thousands of calls, but have to route all outgoint (pstn) calls to a single asterisk/freeswitch server that's actually registered with my provider, wouldn't I loose all my redundancy and concurrency capability??
Is there a better way??
thanks in advance
On 2013-06-19 13:19, Stoyan Mihaylov wrote:
It depends.On Wed, Jun 19, 2013 at 7:30 PM, Jose Suero <ms@mstn.com [3]> wrote:
I can imagine next scenarios:
1. Under SIP trunks you mean calls from your provider to you
A) In case your provider can send calls to you - then you can use
Kamailio, accepting all calls from your provider - based on IP.
B) In case your provider expects registration from your system - then,
at least I - dont know how to do only with Kamailio - Asterisk can
register easily to every provider.
2. Under SIP trunks you mean calls from you to World through your
provider.
A) Your provider can accept all calls from you based on IP - Kamailio
can directly forward calls to your provider.
B) Your provider expects authentication - then again I dont know how
this can be done through Kamailio, but Asterisk can do it easily.
My suggestion is - you can use Kamailio for registration of users and
load balancing, and asterisk servers for everything else.
Hi(havent decided on freeswitch or asterisk) theres a million
Im planning to set kamailio in front of an farm of pbx servers
tutorials on how to do this, what I havent found is what part of myWhats the best practice when It comes to this?
setup actually handles the sip trunks my phone company provides me
with.
sr-users@lists.sip-router.org [1]
Is kamailio going to be receiving the calls from the trunk and
passing them to the PBX or is it the other way around?
please advice
Thanks in advance
Jose Suero
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users [2]
Links:
------
[1] mailto:sr-users@lists.sip-router.org
[2] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[3] mailto:ms@mstn.com
_______________________________________________
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