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


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




A2Billing and OpenSIPS – Part 1

Last updated on 

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.

Last updated by .

조회 수 :
43080
등록일 :
2014.03.04
21:49:32 (*.251.139.148)
엮인글 :
http://webs.co.kr/index.php?document_srl=38911&act=trackback&key=4e4
게시글 주소 :
http://webs.co.kr/index.php?document_srl=38911
List of Articles
번호 제목 글쓴이 조회 수sort 추천 수 날짜
82 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 41439   2014-08-13
 
81 Real-time Charging System for Telecom & ISP environments admin 40898   2014-08-23
 
80 A lightweight RPC library based on XML and HTTP admin 40857   2014-08-18
 
79 A Survey of Open Source Products for Building a SIP Communication Platform admin 40713   2014-10-18
 
78 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 40677   2014-08-24
 
77 UAC Registrant Module admin 40067   2014-09-28
 
76 [Sipdroid] SIP data collection study tour admin 40061   2014-08-23
 
75 MediaProxy Installation Guide admin 39998   2014-08-10
 
74 OpenSER_from_an_asterisk_POV file admin 39838   2013-04-06
 
73 rtpproxy Module admin 39782   2014-03-06
 
72 OpenSIPS Kick Start‎: VIDEO admin 39730   2013-02-20
 
71 CANCEL MESSAGE not handled correctly admin 39723   2014-08-23
 
70 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 39290   2014-08-12
 
69 Opensips install debian admin 39252   2014-03-03
 
68 SigIMS IMS Platform admin 39009   2014-05-24
 
67 fusionPBX install debian wheezy admin 38978   2014-08-09
 
66 OPENSIPS EBOOK admin 38956   2014-08-21
 
65 A2Billing and OpenSIPS config admin 38867   2014-10-20
 
64 opensips-1.10.0_src.tar.gz experimental source code documentation admin 38794   2014-03-09
 
63 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 38370   2014-08-11
 
62 Multimedia Service Platform admin 38177   2014-03-06
 
61 opensips Nat script with RTPPROXY - English Good perfect admin 38040   2014-08-15
 
60 OPENSIP Training VIDEO admin 38023   2013-02-20
 
59 opensips.cfg for Asterisk admin 38020   2014-10-20
 
58 opensips NAT Traversal Module admin 37852   2014-10-02
 
57 kamailio.cfg configuration Example admin 37634   2014-10-04
 
56 OpenSIPS as Homer Capture server admin 37483   2014-08-13
 
55 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 37342   2014-08-12
 
54 Kamailio Nat Traversal using RTPProxy admin 37321   2014-08-11
 
53 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 37266   2014-08-23