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 :
31457
등록일 :
2017.08.29
11:29:22 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311338&act=trackback&key=1dd
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311338
List of Articles
번호 제목 글쓴이 날짜 조회 수sort
171 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 279135
170 Opensips Gateway between SIP and SMPP messages admin 2019-02-19 265931
169 What is new in 1.8.0 opensip admin 2012-05-21 252525
168 What is new in 2.3.0 opensips admin 2017-09-04 244676
167 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 2013-03-31 225708
166 OpenSIPS vs Asterisk admin 2013-04-06 220258
165 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 207898
164 MediaProxy Installation Guide admin 2014-03-06 180458
163 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 180055
162 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 175047
161 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 161815
160 Asterisk v1.4x built on FreeBSD v7.1 UNIX admin 2012-01-06 149576
159 사설 망 환경에서 SIP 의 NAT Traversal 문제 admin 2011-12-23 144177
158 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 143025
157 SIP 트래픽 생성 테스트 툴 admin 2011-12-23 136363
156 opensips command /sbin/opensipsctl detail admin 2017-09-04 124481
155 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 123594
154 Opensips_1.9 install guide this is great I like this admin 2014-03-04 107113
153 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 104759
152 Welcome to the Smartvox Knowledgebase admin 2013-04-06 104305
151 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 101972
150 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 101532
149 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 99676
148 OpenSIPS Control Panel install guide admin 2014-03-06 95595
147 dictionary.opensips radius admin 2017-12-09 94625
146 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 94581
145 My new toy: Bluebox-ng admin 2013-04-06 91088
144 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 90735
143 Conference Support in Kamailio (OpenSER) admin 2014-03-12 84958
142 Kamailo OpenSIPs installation on Debian admin 2014-03-09 81859