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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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111 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 41498   2014-08-10
 
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106 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 37922   2014-08-11
 
105 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 38863   2014-08-12
 
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103 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 41001   2014-08-13
 
102 OpenSIPS as Homer Capture server admin 37022   2014-08-13
 
101 Installation and configuration process record opensips opensips-cp admin 70312   2014-08-13
 
100 OpenSIPS Control Panel (OCP) Installation Guide Good admin 68402   2014-08-13
 
99 opensips Nat script with RTPPROXY - English Good perfect admin 37584   2014-08-15
 
98 A lightweight RPC library based on XML and HTTP admin 40388   2014-08-18
 
97 Opensips Modules Documentation admin 42378   2014-08-18
 
96 Presence Tutorial OpenXCAP setup admin 40844   2014-08-18
 
95 Opensips Documentation Function admin 41212   2014-08-21
 
94 OPENSIPS EBOOK admin 38340   2014-08-21
 
93 Real-time Charging System for Telecom & ISP environments admin 40421   2014-08-23
 
92 Many OPENSIPS Configuration Examples This will Help you admin 49265   2014-08-23
 
91 ICE: The ultimate way of beating NAT in SIP admin 65285   2014-08-23
 
90 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 41548   2014-08-23
 
89 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 36760   2014-08-23
 
88 [Sipdroid] SIP data collection study tour admin 39563   2014-08-23
 
87 CANCEL MESSAGE not handled correctly admin 39278   2014-08-23
 
86 RTPPROXY Admin Guide admin 41524   2014-08-24
 
85 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 40190   2014-08-24
 
84 UAC Registrant Module admin 39543   2014-09-28
 
83 opensips NAT Traversal Module admin 37273   2014-10-02
 
82 kamailio.cfg configuration Example admin 37149   2014-10-04