한국어

소프트스위치

온누리070 플레이스토어 다운로드
    acrobits softphone
     온누리 070 카카오 프러스 친구추가온누리 070 카카오 프러스 친구추가친추
     카카오톡 채팅 상담 카카오톡 채팅 상담카톡
    
     라인상담
     라인으로 공유

     페북공유

   ◎위챗 : speedseoul


  
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app


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.

조회 수 :
31503
등록일 :
2017.08.29
11:29:22 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311338&act=trackback&key=3a8
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311338
List of Articles
번호 제목 글쓴이 조회 수 추천 수 날짜
111 OpenSIPS 2.3 install admin 24033   2017-09-01
 
110 JsSIP: The JavaScript SIP Library admin 20247   2017-09-01
 
109 WebSocket Transport using OpenSIPS admin 23823   2017-09-01
 
» A2Billing and OpenSIPS – Part 1 admin 31503   2017-08-29
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 ...  
107 A2Billing and OpenSIPS – Part 2 admin 33509   2017-08-29
 
106 A2Billing and OpenSIPS – Part 3 admin 20802   2017-08-29
 
105 OpenSIPS 2.3 philosophy admin 21121   2017-08-17
 
104 The timeline for OpenSIPS 2.3 is admin 21381   2017-08-17
 
103 OpenSIPS Control Panel and Homer integration admin 42532   2017-08-17
 
102 Opensips sip capture re designed admin 20754   2017-07-16
 
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 32336   2015-04-04
 
100 WebSocket Support in OpenSIPS 2.1 admin 31030   2015-04-04
 
99 OpenSIPS 2.1 (rc) is available, download now! admin 24061   2015-03-22
 
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 32941   2015-02-25
 
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 41583   2014-11-02
 
96 opensips.cfg for Asterisk admin 36878   2014-10-20
 
95 A2Billing and OpenSIPS config admin 37723   2014-10-20
 
94 Jitsi Videobridge meets WebRTC admin 44668   2014-10-18
 
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 39615   2014-10-18
 
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 35294   2014-10-14
 
91 Opensips TM module enables stateful processing of SIP transactions admin 44971   2014-10-04
 
90 kamailio.cfg configuration Example admin 36559   2014-10-04
 
89 opensips NAT Traversal Module admin 36658   2014-10-02
 
88 UAC Registrant Module admin 38905   2014-09-28
 
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 39562   2014-08-24
 
86 RTPPROXY Admin Guide admin 40965   2014-08-24
 
85 CANCEL MESSAGE not handled correctly admin 38666   2014-08-23
 
84 [Sipdroid] SIP data collection study tour admin 38950   2014-08-23
 
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 36094   2014-08-23
 
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 40963   2014-08-23