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https://sysadminman.net/blog/2013/a2billing-and-opensips-part-1-4775


This is to confirm that SysAdminMan no longer offers FreePBX or A2Billing hosting.There were a few reasons for this decision but one of that main ones was, in my opinion, Sangoma’s aggressive commercialisation of FreePBX and their “FreePBX” trademark. It did not make commercial sense to continue building a business under these circumstances.According to Google Analytics there are still a couple of thousand visitors a week that use the site, so I will leave it here, but will not be adding new guides or tips.


This is part 1 of a 3 part post discussing A2Billing and OpenSIPS. A2Billing is a billing platform for Asterisk, and OpenSIPS is an Open Source SIP Server. In this first part I’m going to talk about what OpenSIPS is and why you may want to use it. In the second part I’ll talk about some prerequisites for the setup I’m going to show, and in the third part will be the OpenSIPS config.

A2Billing works perfectly well without OpenSIPS, so why would you want to use them together? Well, with OpenSIPS sitting in front of A2Billing/Asterisk and handling all of the SIP connections it can provide the following benefits –

  • load balance across multiple Asterisk/A2Billing servers
  • failover – take an Asterisk server out of the cluster if it should fail
  • limit SIP connections so that only the OpenSIPS server talks to Asterisk/A2Billing over SIP
  • register all of your SIP customers in a single place – the OpenSIPS server (the config I show is not going to cover SIP registrations)
  • OpenSIPS has much better logging of SIP connections (than Asterisk) so we can use fail2ban more efficiently to block attacks

There are probably many more benefits than those listed above. OpenSIPS has lots of modules that provide flexibility to handle the SIP connections exactly as you need.

In the config that follows I am going to show how to do SIP termination. SIP clients authenticate to OpenSIPS using either IP or USER/SECRET authentication and then calls are passed to A2Billing/Asterisk for completion. This example does not cover SIP registrations or incoming DID numbers.

OpenSIPS will sit between the A2Billing SIP customers and the A2Billing/Asterisk server. All customer SIP connections will be to the OpenSIPS server, which will then pass these on to Asterisk/A2Billing once authenticated. A2Billing/Asterisk will talk to the call provider directly (not via OpenSIPS). So the setup looks something like this –

A2Billing SIP Customer  -->  OpenSIPS  -->  A2Billing/Asterisk  --> Call provider
                                       -->  A2Billing/Asterisk  --> Call provider
                                       -->  A2Billing/Asterisk  --> Call provider

This diagram above shows calls going to 3 different A2Billing/Asterisk servers. In the example config there is just one set up, but it will be obvious how to add more.

Also, in OpenSIPS there are 2 different ‘load balancing’ modules. There is one called ‘dispatcher’ which in unintelligent and just send the calls to a group of A2Billing/Asterisk servers. And there is a module called ‘load-balancer’ which knows the state of each A2Billing/Asterisk server and evenly distributes the load across them. For simplicity in this example I will be using the ‘dispatcher’ module.

This guide assumes that you have –

  • a working A2Billing/Asterisk server in place
  • a working OpenSIPS v1.8 server in place
  • created a database called ‘opensips’ (as per the OpenSIPS install instructions) that is on MySQL running on the A2BIlling/Asterisk server

We are going to have both the A2Billing and OpenSIPS databases running on the A2Billing server so that we can integrate the two

In part 2 I’ll discuss some of the prerequisites and the database setup.

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11:29:22 (*.160.88.18)
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List of Articles
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112 Using TLS in OpenSIPS v2.2.x admin 53779   2017-09-14
 
111 How to install OpenSIPS on CentOS Debian etc admin 53507   2014-03-05
 
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109 SIPSorcery admin 53164   2014-03-18
 
108 OpenSIPS Installation Notes admin 52991   2014-08-09
 
107 Open Source VOIP applications, both clients and servers. admin 52949   2013-11-20
 
106 Asterisk Installation Asterisk Realtime configuration admin 52477   2013-04-06
 
105 The SIP Router Project admin 52154   2013-04-06
 
104 Where to check OpenSIPS does not start? admin 51722   2014-03-09
 
103 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 51280   2014-03-12
 
102 Building Telephony Systems with OpenSIPS 1.6 books file admin 51094   2014-03-06
 
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100 A2Billing and OpenSIPS admin 50883   2014-03-04
 
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98 Jitsi Videobridge meets WebRTC admin 50716   2014-10-18
 
97 The FreeRADIUS Project admin 50698   2011-12-14
 
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87 Opensips Modules Documentation admin 46765   2014-08-18
 
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84 RTPPROXY Admin Guide admin 46106   2014-08-24
 
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