[SR-Users] Help

SamyGo govoiper at gmail.com
Sun Feb 28 16:50:58 CET 2016


I think the best guide closest to your description is here :

Here is what you need to do. (*Besides mentioning what you tried and what
problems were faced*).

1 - Configure kamailio to use the DB schema where your users are stored
with their password and PBX to use info.
2 - Point Phones to REGISTER to your kamailio.
3 - When a User makes a call execute query in Kamailio to find what client
this user belongs to  and what Asterisk it should be routed to.
4 - Send Calls to the selected Asterisk.

You can further add Memcache to save your DB query in step-3.

Good Part:
1 - Kamailio Authenticates all users and calls.
2 - One Public IP to point all domains/PBX tenants to.
3 - Use RTPproxy to bridge media to Asterisks and you can shift your
Asterisks on Private Subnet too.(depends on your design)
4 - Sending a hand crafted REGISTER to Asterisk makes asterisk aware of the
device state and hence BLF/MWI are handled by Asterisk.

Less Good Part: (As I see it)

Kamailio sends REGISTER packet to just one Asterisk ! thereby only one
server out of pool is aware of the device states. It can be resolved by
extra effort required as following:

      a) Yes we can use Dispatcher and send to failover/loadbalanced
asterisks in the pool
      b) A script of some sorts can be written and started in asterisk
servers to share device states/hints and \
          hence all asterisk servers in pool know whats going on. (I
haven't tried it myself)
      c) REGFWD route can be blocked and BLF, MWI are handled solely by
Kamailio.  (* I personally had rough time with this mostly due to different
standards *from IP Phones)

I'd love to hear other valuable suggestions and experiences.

Hello Kevin,
If I understood properly you want to build a system which authenticates
users and routes the Asterisk servers for communication.

First, Kamailio supports the routing, balancing and authentication. For
example we use Kamailio and Freeswitch. Here the how its work:
We have 1 Kamailio server that makes routes and balancing issues.
First client goes to our Kamailio servers:

*Client -> Internet -> Kamailio (authentication) (address, asked for

After that, Kamailio looks the Freeswitch servers, which is free for

*                  (sending)*
*Kamailio -----------------> Freeswitch Server*
*                  (user req)*

After routing proccess, Kamailio fade from the scene and clients start
communicate with themselves via Freeswitch servers.

BTW, our Freeswitch servers and Kamailio servers stay on different servers.
Of course you can serve on same server too.
If I understood properly, you can do it like this. If I did not, you can
give more details for understanding :)


From: kfpelletier at connextek.ca
To: sr-users at lists.sip-router.org; sr-dev at lists.sip-router.org;
buisness at lists.kamailio.org
Date: Fri, 26 Feb 2016 15:35:50 -0500
Subject: [SR-Users] Help


I work for a VOIP service provider, and have been tasked with optimizing
our infrastructure.  We have been providing VOIP services to our clients
via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a datacenter.
We are looking for some kind of SIP Router, which would authenticate
clients and route their SIP traffic to the appropriate server.  By doing
so, we are hoping to further secure our infrastructure and to possibly have
only one Public IP (which would resolve to the Private IP of the SIP
router).  The Asterisk servers serve
would therefore route all SIP traffic between the phones and the Asterisk
servers, ad the phones would register to the SIP Router.  I have tried many
solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to
configure these services to work the way we want them to.  I am including a
chart along with this email to outline what we would like to accomplish.

Any suggestions or guides would be immensely appreciated.

Thank you all for your time.

*Kevin Farrell Pelletier - Technicien informatique*

TI* //* Réseautique // Téléphonie IP *//* Programmation
IT // Networking // VoIP // Application development

9060, boul Parkway, Anjou, Québec, Canada H1J1N5
* Téléphone / Phone :* 514.907.2000 Ext.203
* F :* 1.888.582.4001  -  *SF/TF :* 1.855.907.2001* Web : *www.connextek.ca

* Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire *
*            Consider the ENVIRONMENT before printing*

_______________________________________________ SIP Express Router (SER)
and Kamailio (OpenSER) - sr-users mailing list sr-users at lists.sip-router.org

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sr-users at lists.sip-router.org
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