Hi
I found this info very helpful when setting up a scenario similar to your
required setup.
https://txlab.wordpress.com/2012/10/21/kamailio-as-a-pass-through-proxy/
Gerry Kernan
Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin
D18 E3C8 | Ireland
Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.ker...@infinityit.ie
Managed IT Services Infinity IT - www.infinityit.ie
IP Telephony Asterisk Consulting – www.asteriskconsulting.com
Contact Centre Total Interact – www.totalinteract.com
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
SamyGo
Sent: Sunday 28 February 2016 15:51
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List <sr-users@lists.sip-router.org>
Subject: Re: [SR-Users] Help
Hi,
I think the best guide closest to your description is here :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
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.
Regards,
Sammy
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
communication)
After that, Kamailio looks the Freeswitch servers, which is free for routing.
(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 :)
Regards.
Barış.
From: kfpellet...@connextek.ca
To: sr-users@lists.sip-router.org; sr-...@lists.sip-router.org;
buisn...@lists.kamailio.org
Date: Fri, 26 Feb 2016 15:35:50 -0500
Subject: [SR-Users] Help
Hi,
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 IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The
Sip Router 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
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