13 Sep
2016
13 Sep
'16
3:11 p.m.
it is many-many examples of kamialio.cfg at the internet that describes
same logic with different staff (like kamailio as registrar and also as
kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me
to understand all it.
really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime-integration-wi…
https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio-3-1-realtime-i…
All this just one of the many variants how you can to integrate it.
Good Luck. I suppose you will know many new cool things when open kamailio
for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery <gr.sabery(a)gmail.com>om>:
For testing purpose you can use example config file it
is a very good
place to start. Also if you want automatic installation and deployment you
can use this project:
https://github.com/ghrst/Kamailio-HA
On Tue, Sep 13, 2016 at 8:57 PM, Valter Nogueira <valter(a)fastway.com.br>
wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" <anfecora(a)gmail.com> escreveu:
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Valter i wouldnt take fully asterisk from the
picture you can use it to
handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" <valter(a)fastway.com.br>
wrote:
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I use Asterisk for SIP and Media Proxy. Despite
the fact that Asterisk
is not a SIP Proxy at all.
Customer registers in a SIP account, sends the invite and thru de
context Asterisk dials out thru a SIP Trunk. Asterisk does the media proxy,
since customer can't route directly to the SIP Trunk (altough it has a
valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out
numbers, keep track of ongoing call, control SIP dialog, retransmit correct
hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio
Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth,
dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg
and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
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