19 Jun
2013
19 Jun
'13
7:30 p.m.
Hi
I'm planning to set kamailio in front of an farm of pbx servers
(haven't decided on freeswitch or asterisk) there's a million tutorials
on how to do this, what I haven't found is what part of my setup
actually handles the sip trunks my phone company provides me with.
What's the best practice when It comes to this?
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
19 Jun
19 Jun
8:19 p.m.
It depends.
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.
On Wed, Jun 19, 2013 at 7:30 PM, Jose Suero <ms(a)mstn.com> wrote:
Hi
I'm planning to set kamailio in front of an farm of pbx servers (haven't
decided on freeswitch or asterisk) there's a million tutorials on how to do
this, what I haven't found is what part of my setup actually handles the
sip trunks my phone company provides me with.
What's the best practice when It comes to this?
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
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http:/…
25 Jun
25 Jun
8:24 a.m.
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.
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.
On Wed, Jun 19, 2013 at 7:30 PM, Jose Suero <ms(a)mstn.com [3]> wrote:
Hi
Im planning to set kamailio in front of an farm of pbx servers
(havent decided on freeswitch or asterisk) theres a million
tutorials on how to do this, what I havent found is what part of my
setup actually handles the sip trunks my phone company provides me
with.
Whats the best practice when It comes to this?
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
sr-users(a)lists.sip-router.org [1]
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
5:55 p.m.
Surely there must be a better way.
For example, if you are a good client for your provider, you can ask him to
forward all calls to your IP. This way, Kamailio will accept all calls and
forward them to multiple Asterisk boxes for processing.
Also - you can put one Asterisk to do only one thing - accept all calls
(from provider), and forward them (to other Asterisk boxes, or to
Kamailio).
May be there are better ways, but in this moment, I cannot imagine more.
PS
There are many providers, and some of them can forward calls. It is good
idea to see if your provider can or cannot really forward calls to your IP.
On Tue, Jun 25, 2013 at 8:24 AM, Jose Suero <ms(a)mstn.com> wrote:
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.
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.
On Wed, Jun 19, 2013 at 7:30 PM, Jose Suero <ms(a)mstn.com [3]> wrote:
Hi
______________________________**_________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http:/…
Im planning to set kamailio in front of an farm of pbx servers
(havent decided on freeswitch or asterisk) theres a million
tutorials on how to do this, what I havent found is what part of my
setup actually handles the sip trunks my phone company provides me
with.
Whats the best practice when It comes to this?
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
sr-users(a)lists.sip-router.org [1]
http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http:/…
Links:
------
[1] mailto:sr-users@lists.sip-**router.org<sr-users@lists.sip-router.org>
[2]
http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http:/…
[3] mailto:ms@mstn.com
4171
days inactive
4177
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3 comments
2 participants
participants (2)
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Jose Suero -
Stoyan Mihaylov