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 :
32685
등록일 :
2017.08.29
11:29:22 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311338&act=trackback&key=dd2
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311338
List of Articles
번호 제목 글쓴이 조회 수sort 추천 수 날짜
112 Opensips TM module enables stateful processing of SIP transactions admin 46204   2014-10-04
 
111 Using TLS in OpenSIPS v2.2.x admin 46179   2017-09-14
 
110 How to install OpenSIPS on CentOS Debian etc admin 46179   2014-03-05
 
109 The SIP Router Project admin 46172   2013-04-06
 
108 Jitsi Videobridge meets WebRTC admin 45906   2014-10-18
 
107 Open Source VOIP applications, both clients and servers. admin 45387   2013-11-20
 
106 opensips 1.11.2 install guide good 인스톨 가이드 admin 45129   2014-08-09
 
105 The FreeRADIUS Project admin 45059   2011-12-14
 
104 SIPSorcery admin 44890   2014-03-18
 
103 OpenSIPS Module Interface admin 44212   2017-12-07
 
102 Where to check OpenSIPS does not start? admin 44034   2014-03-09
 
101 opensips complete configuration example admin 43826   2017-12-10
 
100 OpenSIPS Control Panel and Homer integration admin 43729   2017-08-17
 
99 List of SIP response codes admin 43674   2017-12-20
 
98 MediaProxy wiki page install configuration admin 43670   2014-08-11
 
97 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 43645   2014-03-12
 
96 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 43624   2014-03-12
 
95 Building Telephony Systems with OpenSIPS 1.6 books file admin 43468   2014-03-06
 
94 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 43456   2017-09-04
 
93 The Impact of TLS on SIP Server Performance file admin 43078   2014-03-12
 
92 A2Billing and OpenSIPS admin 43035   2014-03-04
 
91 Opensips Modules Documentation admin 42825   2014-08-18
 
90 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 42696   2014-11-02
 
89 rfc5766-turn-server admin 42266   2013-03-21
 
88 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 41982   2014-08-23
 
87 RTPPROXY Admin Guide admin 41939   2014-08-24
 
86 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 41908   2014-08-10
 
85 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 41679   2014-03-07
 
84 Opensips Documentation Function admin 41651   2014-08-21
 
83 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 41428   2014-08-13