한국어

소프트스위치

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

     페북공유
    
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app
■ 해외 로잉 무료 스마트폰 휴대폰 070 인터넷폰 인터넷전화 국내 해외 가입 상사 주재원 교민 유학생 여행 등 ■

https://sysadminman.net/blog/2013/a2billing-and-opensips-part-3-4814



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 post has the actual config of OpenSIPS described in part 1 and part 2. Some of this config will not make sense unless you read those parts.

First a warning … many, many people want to use your call credit!! Make sure that your systems are secure. If only the OpenSIPS server needs to talk to your A2Billing/Asterisk servers over SIP then use a fierwall to block other connections.

In the configuration below OpenSIPS does not handle the Audio/RTP traffic, this is passed directly to the Asterisk/A2Billing server.

The code below is the whole opensips.cfg file, just broken up with some description. All indentation has been removed, apologies if this sometimes makes it difficult to read.

First some global settings, including the IP address of the OpenSIPS server –

listen=udp:1.1.1.1:5060 # CUSTOMIZE ME
debug=1
log_stderror=no
log_facility=LOG_LOCAL6
fork=yes
children=4
dns_try_ipv6=no
auto_aliases=no
disable_tcp=yes
disable_tls=yes
server_signature=no

next we load the modules that are required –

mpath="/usr/local/lib/opensips/modules/"
loadmodule "signaling.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "sipmsgops.so"
loadmodule "mi_fifo.so"
loadmodule "uri.so"
loadmodule "db_mysql.so"
loadmodule "avpops.so"
loadmodule "acc.so"
loadmodule "dispatcher.so"
loadmodule "permissions.so"
loadmodule "dialog.so"
loadmodule "siptrace.so"
loadmodule "auth.so"
loadmodule "auth_db.so"

now we set the database URL for the modules that are going to use our MySQL database. This database is on the A2Billing server –

modparam("acc|dispatcher|permissions|dialog|siptrace|auth_db|avpops","db_url","mysql://username:password@2.2.2.2/opensips")

now we set some other module options –

modparam("tm", "fr_inv_timer", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
modparam("rr", "append_fromtag", 0)
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
modparam("uri", "use_uri_table", 0)
modparam("acc", "db_flag", 1)
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_probing_mode", 0)
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "ds_ping_interval", 30)
modparam("dialog", "db_mode", 1)
modparam("dialog", "dlg_match_mode", 2)
modparam("siptrace", "trace_on", 0)
modparam("siptrace", "trace_flag", 22)
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "skip_version_check", 1)

and now to the actual routing. The first block has some general default code that rejects some packets and also immediately relays sip dialogs that have already been established –

route{
if (!mf_process_maxfwd_header("10") && $retcode==-1) {
sl_send_reply("483","Too Many Hops");
exit;
}
if (has_totag()) {
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
record_route();
}
route(RELAY);
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}

next more generic checks. Relay CANCEL messages and reject others with incomplete URIs –

if (is_method("CANCEL")) {
if (t_check_trans())
t_relay();
exit;
} else if (!is_method("INVITE")) {
send_reply("405","Method Not Allowed");
xlog("$rm FAILED: $si / $ct / $fu\n");
exit;
}
if ($rU==NULL) {
sl_send_reply("484","Address Incomplete");
exit;
}

next drop some more invalid packets –

if (loose_route()) {
xlog("L_ERR","Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}

next we are going to create a route header on the packet and switch on some accounting –

record_route();
setflag(1);

now we are going to choose the Asterisk server that the call is passed to. This uses the ds_select_dst command which is part of the dispatcher module. The available Asterisk servers are selected from the ‘dispatcher’ table in the MysQL database. The ‘1’ relates to the ‘setid’ field in the ‘dispatcher’ table, and the ‘destination’ field should be in the format ‘sip:2.2.2.2:5060’ –

if ( !ds_select_dst("1","0") ) {
send_reply("500","No Destination available");
xlog("$rm FAILED: NO DESTINATION: $si / $tu / $ru\n");
exit;
}
t_on_failure("GW_FAILOVER");

Now the good stuff! This is where we do IP authentication, and route the call if it is valid. We use the ‘check_source_address’ command which is part of the permissions module. This is going to look in the MySQL database for a matching IP address. The “1” is a group ID that we hard coded when we set up the VIEW in MySQL. If the IP address matches then the account code is returned to us (because we stored it in the context_info field in MySQL), and we set this to the variable “$avp(accountcode”. We then set this variable in a header on the SIP INVITE packet and send it to A2Billing. In part 2 I showed how to extract this SIP header and set the account code in the Asterisk dialplan, so that A2Billing knows which customer to charge for the call –

if (check_source_address("1","$avp(accountcode)")) {
xlog("L_INFO", "IP $rm DIALLED: $si / $ru / Accountcode: $avp(accountcode) ");
remove_hf("P-Accountcode");
append_hf("P-Accountcode: $avp(accountcode)\r\n");
route(RELAY);
};

next, if IP authentication above failed we want to challenge the caller (the IP address sending the SIP INVITE) for a Username and Secret. “subscriber” is the name of the VIEW in MySQL where we are storing the customer credentials. These are picked up straight from the A2BIlling customers SIP account –

if (!proxy_authorize("", "subscriber")) {
$var(debug) = proxy_authorize("", "subscriber");
xlog("Not Proxy Authorize: $var(debug)");
proxy_challenge("", "0");
exit;
}

now, if the customer passed the authorisation above, we want to send the call to our Asterisk/A2billing server. First though we want to set the Account Code so that A2Billing knows which customer to charge the call to. This is similar to what we did for IP authentication, but we need to run a separate command to retrieve the account code from the ‘rpid’ field in the ‘subscriber’ table where we stored it. $au is the authorized username –

avp_db_query("select rpid from subscriber where username='$au'","$avp(accountcode)");
remove_hf("P-Accountcode");
append_hf("P-Accountcode: $avp(accountcode)\r\n");
xlog("L_INFO", "AUTH $rm DIALLED: $si / $ru / Accountcode: $avp(accountcode) ");
consume_credentials();
route(RELAY);

}

Finally, we called route(RELAY) several times in the script above, and we define that here, and a couple of other bits, we forward the packets with t_relay –

route[RELAY] {
if (!t_relay()) {
sl_reply_error();
};
exit;
}

failure_route[GW_FAILOVER] {
if (t_was_cancelled()) {
exit;
}
if (t_check_status("(408)|([56][0-9][0-9])")) {
xlog("Failed trunk $rd/$du detected \n");
if ( ds_next_dst() ) {
t_on_failure("GW_FAILOVER");
t_relay();
exit;
}
send_reply("500","All GW are down");
}
}

And that’s the end of the config file!

The code above is definitely not designed to be totally cut and paste. You are going to have to check some documentation and have a fair understanding of what’s going on and how the call is being handled. I would also suggest learning the xlog command and some of the variables available. This you can use at various points in the script to log some output and see why you calls might be failing.

If anyone experienced with OpenSIPS (or Kamailio) can offer any suggestions for how to improve the config I’d be interested to hear them. Also, of anyone with Kamailio experience could let me know how different that config would look in that I’d be interested to hear that too. Thanks!


조회 수 :
2655
등록일 :
2017.08.29
11:27:15 (*.160.88.18)
엮인글 :
http://webs.co.kr/index.php?document_srl=3311334&act=trackback&key=565
게시글 주소 :
http://webs.co.kr/index.php?document_srl=3311334
List of Articles
번호 제목 글쓴이 날짜 조회 수
158 List of SIP response codes admin 2017-12-20 1126
157 opensips/modules/event_routing/ Push Notification Call pickup admin 2017-12-20 802
156 opensips push notification How to detail file admin 2017-12-20 783
155 OpenSIPS routing logic admin 2017-12-12 897
154 OpenSIPS example configuration admin 2017-12-12 862
153 opensips log output admin 2017-12-11 823
152 opensips complete configuration example admin 2017-12-10 864
151 Opensips1.6 ebook detail configuration and SIP signal and NAT etc file admin 2017-12-10 938
150 dictionary.opensips radius admin 2017-12-09 1291
149 what is record_route() in opensips ? admin 2017-12-09 1316
148 what is loose_route() in opensips ? file admin 2017-12-09 1352
147 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 1317
146 in opensips db_does_uri_exist() what is admin 2017-12-09 1284
145 in opensips what is has_totag() admin 2017-12-09 1310
144 opensips exec module admin 2017-12-08 1367
143 opensips push notification How to admin 2017-12-07 1341
142 OpenSIPS Module Interface admin 2017-12-07 1391
141 opensips configuration config explain easy basic 오픈쉽스 컨피그레이션 기본 설명 file admin 2017-12-07 1373
140 openssl 을 이용한 인증서 생성 절차를 정리한다. 개인키 CSR SSL 인증서 파일 생성 admin 2017-09-14 2254
139 Documentation -> Tutorials -> TLS opensips.cfg admin 2017-09-14 2288
138 Using TLS in OpenSIPS v2.2.x admin 2017-09-14 2231
137 opensips tls cfg admin 2017-09-14 2384
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 2514
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 2017-09-13 2265
134 OpenSIPS command line tricks admin 2017-09-13 2232
133 Fail2Ban Freeswitch How to secure admin 2017-09-12 2323
132 opensips.cfg. sample admin 2017-09-12 2249
131 Advanced SIP scenarios with Event-based-Routing admin 2017-09-11 2340
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 2439
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 2017-09-09 3013
128 rtpengine config basic and opensips configuration and command admin 2017-09-06 2382
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 2017-09-06 2330
126 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 2391
125 rtpengine install and config admin 2017-09-05 2415
124 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 2469
123 compile only the textops module make modules=modules/textops modules admin 2017-09-05 2410
122 opensips command /sbin/opensipsctl detail admin 2017-09-04 2455
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 2017-09-04 2470
120 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 2375
119 What is new in 2.3.0 opensips admin 2017-09-04 2691
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 2017-09-04 2418
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 2644
116 Using TLS in OpenSIPS v2.2.x configuration admin 2017-09-04 2495
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 2017-09-02 2494
114 You can install CDRTool in the following ways: admin 2017-09-01 2564
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 2017-09-01 2569
112 Opensips 2.32 download admin 2017-09-01 2450
111 OpenSIPS 2.3 install admin 2017-09-01 2644
110 JsSIP: The JavaScript SIP Library admin 2017-09-01 2519
109 WebSocket Transport using OpenSIPS admin 2017-09-01 2592
108 A2Billing and OpenSIPS – Part 1 admin 2017-08-29 2576
107 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 2549
» A2Billing and OpenSIPS – Part 3 admin 2017-08-29 2655
105 OpenSIPS 2.3 philosophy admin 2017-08-17 2907
104 The timeline for OpenSIPS 2.3 is admin 2017-08-17 3106
103 OpenSIPS Control Panel and Homer integration admin 2017-08-17 2694
102 Opensips sip capture re designed admin 2017-07-16 2849
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 2015-04-04 7542
100 WebSocket Support in OpenSIPS 2.1 admin 2015-04-04 8076
99 OpenSIPS 2.1 (rc) is available, download now! admin 2015-03-22 7596
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 2015-02-25 9155
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 2014-11-02 16856
96 opensips.cfg for Asterisk admin 2014-10-20 19002
95 A2Billing and OpenSIPS config admin 2014-10-20 18356
94 Jitsi Videobridge meets WebRTC admin 2014-10-18 18500
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 2014-10-18 18018
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 2014-10-14 18225
91 Opensips TM module enables stateful processing of SIP transactions admin 2014-10-04 15923
90 kamailio.cfg configuration Example admin 2014-10-04 18102
89 opensips NAT Traversal Module admin 2014-10-02 17423
88 UAC Registrant Module admin 2014-09-28 19079
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 2014-08-24 18094
86 RTPPROXY Admin Guide admin 2014-08-24 18477
85 CANCEL MESSAGE not handled correctly admin 2014-08-23 18323
84 [Sipdroid] SIP data collection study tour admin 2014-08-23 18907
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 2014-08-23 18922
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 2014-08-23 18770
81 ICE: The ultimate way of beating NAT in SIP admin 2014-08-23 18502
80 Many OPENSIPS Configuration Examples This will Help you admin 2014-08-23 18145
79 Real-time Charging System for Telecom & ISP environments admin 2014-08-23 18775
78 OPENSIPS EBOOK admin 2014-08-21 18828
77 Opensips Documentation Function admin 2014-08-21 18949
76 Presence Tutorial OpenXCAP setup admin 2014-08-18 18050
75 Opensips Modules Documentation admin 2014-08-18 18894
74 A lightweight RPC library based on XML and HTTP admin 2014-08-18 18386
73 opensips Nat script with RTPPROXY - English Good perfect admin 2014-08-15 16534
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 2014-08-13 16409
71 Installation and configuration process record opensips opensips-cp admin 2014-08-13 35162
70 OpenSIPS as Homer Capture server admin 2014-08-13 16240
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 2014-08-13 18245
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 2014-08-12 18660
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 2014-08-12 17621
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 2014-08-11 18961
65 Kamailio Nat Traversal using RTPProxy admin 2014-08-11 18538
64 MediaProxy wiki page install configuration admin 2014-08-11 18560
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 27929
62 MediaProxy Installation Guide admin 2014-08-10 18137
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 2014-08-10 19296
60 Opensips Installation, How to. Good guide wiki page admin 2014-08-10 16172
59 OpenSIPS Installation Notes admin 2014-08-09 15489
58 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 18426
57 opensips 1.11.2 install Good Giide admin 2014-08-09 18480
56 fusionPBX install debian wheezy admin 2014-08-09 18390
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 2014-08-09 18059
54 SigIMS IMS Platform admin 2014-05-24 19075
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 2014-04-05 20853
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 21350
51 SIPSorcery admin 2014-03-18 19371
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 19733
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 22292
48 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 18090
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 20268
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 17452
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 19562
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 18533
43 Where to check OpenSIPS does not start? admin 2014-03-09 18998
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 19895
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 18550
40 Using the openSIPS Registrant Module admin 2014-03-09 20007
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 17911
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 19455
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 24015
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 17798
35 OpenSIPS Control Panel install guide admin 2014-03-06 18735
34 rtpproxy Module admin 2014-03-06 19416
33 MediaProxy Installation Guide admin 2014-03-06 20565
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 20036
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 19549
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 20409
29 Multimedia Service Platform admin 2014-03-06 18823
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 19813
27 Opensips Installation, How to. admin 2014-03-05 16238
26 100% CPU usage opensips admin 2014-03-05 19159
25 A2Billing and OpenSIPS admin 2014-03-04 20093
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 24626
23 Opensips install debian admin 2014-03-03 20144
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 20534
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 21162
20 My new toy: Bluebox-ng admin 2013-04-06 34812
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 30670
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 23971
17 The SIP Router Project admin 2013-04-06 22933
16 Kamailio :: A Quick Introduction admin 2013-04-06 20148
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 20786
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 24629
13 OpenSIPS vs Asterisk admin 2013-04-06 41533
12 OpenSER_from_an_asterisk_POV file admin 2013-04-06 20731
11 Using SIP Devices behind NAT OPensip Asterisk IPPhone SIP Telephony file admin 2013-03-31 42906
10 rfc5766-turn-server admin 2013-03-21 22609
9 OpenSIPS Kick Start‎: VIDEO admin 2013-02-20 20026
■ 해외 로잉 무료 스마트폰 휴대폰 070 인터넷폰 인터넷전화 국내 해외 가입 상사 주재원 교민 유학생 여행 등 ■