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http://wiki.disorder.sk/howto:sip_pbx_-_opensips_and_asterisk_configuration


http://www.slideshare.net/saghul/opensips-workshop



SIP PBX - OpenSIPS and Asterisk configuration

This document explains how to install and configure Asterisk 1.6 PBX and OpenSIPS (OpenSER or Kamailio will probably work too).

OpenSIPS will handle registrations and SIP/SIMPLE IM and presence. Asterisk will do everything else. This will not scale well but OpenSIPS configuration can be modified to not route all calls through Asterisk but this will make some services unavailable.

Configuration in this document implements:

  • Instant Messaging and Presence
  • Conference bridge (extensions starting with 9)
  • VoiceMail (777778)
  • Announcements – date, time (600601)
  • Music On Hold
  • Unconditional and Busy forwarding (*21*61)
  • Redial (*5)

Latter three services requires routing through Asterisk (forwarding and redial could be implemented within OpenSIPS).

We'll be using one host with IP address 10.0.0.105. OpenSIPS will run listen on port 5060 and Asterisk on 50600. Don't forget to alter your firewall.

SIP clients

Tested with these SIP clients (both can be really weird sometimes):

You can test only with Ekiga on the same host. Start gconf-editor and change SIP port (e.g. 5059).

Installation

On Debian you can use official Asterisk 1.6 packages and OpenSIPS binaries from repository in Links section. Tested on amd64architecture.

OpenSIPS

OpenSIPS configuration

Note: We will use MySQL database. Uncomment DBENGINE=MYSQL in /etc/opensips/opensipsctlrc.

Warning: This howto was reconstructed from installation notes. Please improvise if some error pops up :)

Initialise the OpenSIPS database with opensipsdbctl create command:

# opensipsdbctl create
MySQL password for root: 
INFO: test server charset
INFO: creating database opensips ...
INFO: Core OpenSIPS tables succesfully created.
Install presence related tables? (y/n): y
INFO: creating presence tables into opensips ...
INFO: Presence tables succesfully created.
Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist? (y/n): n

Create some users (default database password is opensipsrw):

opensipsctl add 10001@10.0.0.105 1111
opensipsctl add 10002@10.0.0.105 1111

Asterisk

Asterisk MySQL tables

We need to alter OpenSIPS tables for Asterisk integration:

mysql -p -u root
USE opensips;
 
-- vmail_password from 8 to 10 in asterisk 1.6
ALTER TABLE subscriber ADD COLUMN `vmail_password` varchar(10) NOT NULL DEFAULT '1';
ALTER TABLE subscriber ADD COLUMN `first_name` varchar(25) NOT NULL DEFAULT '';
ALTER TABLE subscriber ADD COLUMN `last_name` varchar(45) NOT NULL DEFAULT '';
ALTER TABLE subscriber ADD COLUMN `email_address` varchar(50) NOT NULL DEFAULT '';
ALTER TABLE subscriber ADD COLUMN `datetime_created` datetime NOT NULL DEFAULT '0000-00-00 00:00:00';

Now create database for Asterisk:

CREATE DATABASE asterisk;
GRANT ALL PRIVILEGES ON asterisk.* TO 'asterisk' IDENTIFIED  BY 'heslo';

Now create the tables (actually we will not be using the voicemessages table):

USE asterisk;
 
-- create table to store the voicemail massages
CREATE TABLE `voicemessages` (
  `id` int(11) NOT NULL AUTO_INCREMENT,
  `msgnum` int(11) NOT NULL DEFAULT '0',
  `dir` varchar(80) DEFAULT '',
  `context` varchar(80) DEFAULT '',
  `macrocontext` varchar(80) DEFAULT '',
  `callerid` varchar(40) DEFAULT '',
  `origtime` varchar(40) DEFAULT '',
  `duration` varchar(20) DEFAULT '',
  `mailboxuser` varchar(80) DEFAULT '',
  `mailboxcontext` varchar(80) DEFAULT '',
  `recording` longblob,
  PRIMARY KEY  (`id`),
  KEY `dir` (`dir`)
) ENGINE=MyISAM;
 
 
-- create the asterisk users tables as a view over the OpenSIPS subscriber table
CREATE VIEW `asterisk`.`sipusers` AS SELECT
  `opensips`.`subscriber`.`username` AS `name`,
  `opensips`.`subscriber`.`username` AS `defaultuser`,
  _latin1'friend' AS `type`,
  NULL AS `secret`,
  `opensips`.`subscriber`.`domain` AS `host`,
  concat(`opensips`.`subscriber`.`rpid`,_latin1' ',_latin1'<',`opensips`.`subscriber`.`username`,_latin1'>') AS `callerid`,
  _latin1'default' AS `context`,
  `opensips`.`subscriber`.`username` AS `mailbox`,
  _latin1'yes' AS `nat`,
  _latin1'no' AS `qualify`,
  `opensips`.`subscriber`.`username` AS `fromuser`,
  NULL AS `authuser`,
  `opensips`.`subscriber`.`domain` AS `fromdomain`,
  NULL AS `insecure`,
  _latin1'no' AS `canreinvite`,
  NULL AS `disallow`,
  NULL AS `allow`,
  NULL AS `restrictcid`,
  `opensips`.`subscriber`.`domain` AS `defaultip`,
  `opensips`.`subscriber`.`domain` AS `ipaddr`,
  _latin1'05060' AS `port`,
  NULL AS `regseconds`
FROM `opensips`.`subscriber`;
 
 
-- create the asterisk voceimail users table as a view over the OpenSIPS subscriber table
CREATE VIEW `asterisk`.`vmusers` AS SELECT
  concat(`opensips`.`subscriber`.`username`,`opensips`.`subscriber`.`domain`) AS `uniqueid`,
  `opensips`.`subscriber`.`username` AS `customer_id`,
  _latin1'default' AS `context`,
  `opensips`.`subscriber`.`username` AS `mailbox`,
  `opensips`.`subscriber`.`vmail_password` AS `password`,
  concat(`opensips`.`subscriber`.`first_name`,_latin1' ',`opensips`.`subscriber`.`last_name`) AS `fullname`,
  `opensips`.`subscriber`.`email_address` AS `email`,
  NULL AS `pager`,
  `opensips`.`subscriber`.`datetime_created` AS `stamp`
FROM `opensips`.`subscriber`;
 
 
-- create the asterisk voicemail aliases table as a view over the OpenSIPS dbaliases table
CREATE VIEW `asterisk`.`vmaliases` AS SELECT
  `opensips`.`dbaliases`.`alias_username` AS `alias`,
  _latin1'default' AS `context`,
  `opensips`.`dbaliases`.`username` AS `mailbox`
FROM `opensips`.`dbaliases`;
 
 
-- create the meetme database (conference bridge)
CREATE TABLE rooms (
  confno varchar(80) NOT NULL DEFAULT 0,
  pin varchar(20),
  adminpin varchar(20),
  members int NOT NULL DEFAULT 0,
  PRIMARY KEY (confno)
);

Creating MeetMe conference rooms

You can restrict access by setting pin:

INSERT INTO rooms(confno,pin,adminpin) VALUES(99999,1111,1234);
INSERT INTO rooms(confno,adminpin) VALUES(99998,1234);

MeetMe dependencies

MeetMe conference service will need dahdi.ko and dahdi_dummy.ko modules:

modprobe dahdi_dummy

Debian way for creating these modules is:

apt-get install dahdi-source
m-a a-i dahdi-linux

Making Asterisk to use MySQL (realtime)

Install odbcinst and libmyodbc.

/etc/odbcinst.ini:

[MySQL]
Description = MySQL driver
Driver      = /usr/lib/odbc/libmyodbc.so
Setup       = /usr/lib/odbc/libodbcmyS.so
CPTimeout   =
CPReuse	    =
UsageCount  = 1

/etc/odbc.ini

[MySQL-asterisk]
Description = MySQL Asterisk database
Trace       = Off
TraceFile   = stderr
Driver      = MySQL
SERVER      = localhost
USER        = asterisk
PASSWORD    = heslo
PORT        = 3306
DATABASE    = asterisk

/etc/asterisk/res_odbc.conf:

[asterisk]
enabled => yes
dsn => MySQL-asterisk
username => asterisk
password => heslo
pre-connect => yes

/etc/asterisk/extconfig.conf:

[settings]
sipusers => odbc,asterisk,sipusers
sippeers => odbc,asterisk,sipusers
voicemail => odbc,asterisk,vmusers
meetme => odbc,asterisk,rooms

How it works

Registrations, presence and messages go to OpenSIPS server. Extensions prefixed with ”**” are routed directly (with voicemail fallback). Everything else goes to Asterisk.

Services at Asterisk:

  • 1000 – demo, along with other somewhat functional extensions
  • 444 – echo test
  • 600601 – time, date announcements
  • 777 – global voicemail login (access any mailbox)
  • 778 – personal voicemail login (can be altered to bypass auth)
  • VMR_… – voicemail recording
  • 9XXXX – conference bridge
  • 1XXXX – phones (only these are stored for redial)
  • *5 – redial
  • *21*… / #21# – set/reset unconditional forwarding
  • *61*… / #61# – set/reset busy forwarding

Listings

''opensips.cfg''

#
# $Id$
#

# ----------- global configuration parameters ------------------------

debug=3           # debug level (cmd line: -dddddddddd)
fork=yes          # daemonize
#fork=no
log_stderror=no   # (cmd line: -E)
#log_stderror=yes

check_via=no      # (cmd. line: -v)
dns=no            # (cmd. line: -r)
rev_dns=no        # (cmd. line: -R)
children=4

listen=udp:10.0.0.105:5060
#listen=tcp:10.0.0.105:5060

#
# ------------------ module loading ----------------------------------

# Uncomment this if you want to use SQL database
mpath="/usr/lib/opensips/modules"
loadmodule "db_mysql.so"

#loadmodule "nathelper.so"

loadmodule "xlog.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "signaling.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "avpops.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
loadmodule "group.so"
loadmodule "uri.so"

loadmodule "presence.so"
loadmodule "presence_xml.so"
loadmodule "presence_mwi.so"

# -- presence params --
modparam("presence|presence_xml", "db_url", "mysql://opensips:opensipsrw@localhost/opensips")
modparam("presence_xml", "force_active", 1)
modparam("presence", "server_address", "sip:10.0.0.105:5060" )
modparam("presence", "max_expires_publish", 100)
#modparam("presence", "max_expires_subscribe", 140)
modparam("presence", "expires_offset", 5)

modparam("presence", "fallback2db", 1)

loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")

# ----------------- setting module-specific parameters ---------------
modparam("usrloc|auth_db|avpops|group",
    "db_url", "mysql://opensips:opensipsrw@localhost/opensips")

# -- usrloc params --
# persistent storage
modparam("usrloc", "db_mode", 2)

# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config), 
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

modparam("avpops", "avp_table", "usr_preferences")

# -------------------------  request routing logic -------------------

# main routing logic

route{

	# initial sanity checks -- messages with
	# max_forwards==0, or excessively long requests
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	};

	if (msg:len >=  2048 ) {
		sl_send_reply("513", "Message too big");
		exit;
	};

	#initial requests

	# CANCEL processing
	if (is_method("CANCEL")) {
		if (t_check_trans())
			t_relay();
		exit;
	}

	t_check_trans();

	# we record-route all messages -- to make sure that
	# subsequent messages will go through our proxy; that's
	# particularly good if upstream and downstream entities
	# use different transport protocol
	if (!method=="REGISTER")
		record_route();

	# subsequent messages withing a dialog should take the
	# path determined by record-routing
	if (loose_route()) {
		# mark routing logic in request
		append_hf("P-hint: rr-enforced\r\n"); 
		route(1);
	};

	if (!uri==myself) {
		# mark routing logic in request
		append_hf("P-hint: outbound\r\n"); 
		# if you have some interdomain connections via TLS
		#if(uri=~"@tls_domain1.net") {
		#	t_relay("tls:domain1.net");
		#	exit;
		#} else if(uri=~"@tls_domain2.net") {
		#	t_relay("tls:domain2.net");
		#	exit;
		#}
		route(1);
	};

	# if the request is for other domain use UsrLoc
	# (in case, it does not work, use the following command
	# with proper names and addresses in it)
	if (uri==myself) {

		# presence handling
		if( is_method("PUBLISH|SUBSCRIBE"))
			route(2);

		if (method=="REGISTER") {
			# Uncomment this if you want to use digest authentication
			if (!www_authorize("opensips.org", "subscriber")) {
				www_challenge("opensips.org", "0");
				exit;
			};

			save("location");
			exit;
		};

		if (is_method("INVITE")) {
			setflag(1); # do accounting

			if (!uri=~"sip:\*\*") {
				route(3);
			} else {
				strip(2);
			}
		}

		# conference
		if(is_method("INVITE") && $rU=~"^9[0-9]{4}$") {
			xlog("conference\n");
			route(3);
	}

		# special services (3-4 digits)
		if(is_method("INVITE") && $rU=~"^[0-9]{3,4}$") {
			xlog("other service\n");
			route(3);
		}

		# native SIP destinations are handled using our USRLOC DB
		if (!lookup("location")) {
			#sl_send_reply("404", "Not Found");

			# voicemail
			xlog("native fallback to voicemail\n");
			prefix("VMR_");
			rewritehostport("10.0.0.105:50600");
			route(1);

			exit;
		};
		append_hf("P-hint: usrloc applied\r\n"); 
	};

	# when routing via usrloc, log the missed calls also
	setflag(2);

	# arm a failure route in order to catch failed calls
	# targeting local subscribers; if we fail to deliver
	# the call to the user, we send the call to voicemail
	t_on_failure("1");

	route(1);
}


route[1] {
	# send it out now; use stateful forwarding as it works reliably
	# even for UDP2TCP
	if (!t_relay()) {
		sl_reply_error();
	};
	exit;
}

# presence handling route
route[2]
{
	# absorb retransmissions
	if (! t_newtran())
	{
	        sl_reply_error();
	        exit;
	};

	#handle presence requests

	if(is_method("PUBLISH"))
	{
		if($hdr(Sender)!= NULL) {
			handle_publish("$hdr(Sender)");
		} else {
			handle_publish();
			#t_release();
		}
	} else if(is_method("SUBSCRIBE")) {
		handle_subscribe();
		#t_release();
	};

	exit;
}

# asterisk services
route[3] {
	xlog("forwarding to asterisk\n");

	# to asterisk
	rewritehostport("10.0.0.105:50600");

	if (!t_relay()) {
		sl_reply_error();
	};

	exit;
}

failure_route[1] {
	if (t_was_cancelled()) {
		exit;
	}

	# if the failure code is "408 - timeout" or "486 - busy",
	# forward the calls to voicemail recording
	if (t_check_status("486|408")) {
		xlog("L_INFO", "failure_route - call forward to Voice Mail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");

		rewritehostport("10.0.0.105:50600");
		prefix("VMR_");

		t_relay();
		exit;
	}
}

''custom_extensions.conf''

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo

;;; apps context

[default]
include => local

[apps]
include => demo

exten => 444,1,Answer()
;exten => 444,n,SendText(Here is the place where you write the text for your advertisement)
;exten => 444,n,NoOp(===== ${SENDTEXTSTATUS} =====)
exten => 444,n,Wait(1)
exten => 444,n,Echo

; conference
exten => _9XXXX,1,Ringing
exten => _9XXXX,2,Wait(1)
exten => _9XXXX,3,MeetMe(${EXTEN})
exten => _9XXXX,4,Hangup

; voicemail
exten => 777,1,Ringing
exten => 777,n,Wait(1)
exten => 777,n,Answer
exten => 777,n,Wait(1)
exten => 777,n,VoiceMailMain(@default)
exten => 777,n,Hangup

exten => 778,1,Ringing
exten => 778,n,Wait(1)
exten => 778,n,Answer
exten => 778,n,Wait(1)
exten => 778,n,NoOp(=== Mailbox: ${CALLERID(all)} ===)
; add ,s to auth automatically
exten => 778,n,VoiceMailMain(${CALLERID(num)}@default)
exten => 778,n,Hangup

; announcement: time
exten => 600,1,Ringing
exten => 600,2,Wait(1)
;exten => 600,3,SayUnixTime(,Europe/Bratislava,HMp)
exten => 600,3,SayUnixTime(,Europe/Bratislava,HM)
exten => 600,4,Hangup

; announcement:date
exten => 601,1,Ringing
exten => 601,2,SayUnixTime(,Europe/Bratislava,ABdY)
exten => 601,3,Hangup

; voicemail
exten => _VMR_.,1,NoOp(===== EXTENSION --> ${EXTEN} =====)
exten => _VMR_.,n,Ringing
exten => _VMR_.,n,Wait(1)
exten => _VMR_.,n,Answer
exten => _VMR_.,n,Wait(1)
exten => _VMR_.,n,Voicemail(${EXTEN:4}@default,u)
exten => _VMR_.,n,Hangup

; redial
exten = *5,1,Set(temp=${DB(RepeatDial/${CALLERID(num)})})
exten = *5,2,Dial(SIP/**${temp}@10.0.0.105)
;exten = *5,3,NoOp(= ${temp} =)

; Unconditional Call Forward
exten => _*21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten => _*21*X.,2,Hangup
exten => _*21*X.,3,NoOp(=HERE=)
exten => #21#,1,NoOp(${DB_DELETE(CFIM/${CALLERID(NUM)})})
exten => #21#,2,Hangup

; Call Forward on Busy or Unavailable
exten => _*61*X.,1,Set(DB(CFBS/${CALLERID(num)})=${EXTEN:4})
exten => _*61*X.,2,Hangup
exten => #61#,1,NoOp(${DB_DELETE(CFBS/${CALLERID(NUM)})})
exten => #61#,2,Hangup


;;; incoming calls context

[incoming]
;exten => 3221234567,1,Dial(SIP/10001,30)
;exten => 3221234567,n,Dial(SIP/10002,30)
;exten => 3221234567,n,Dial(SIP/10000,30)

;;; outgoing calls context

; local calls only
[local]
include => apps

; dial and mailbox
;exten => _1XXXX,1,Dial(SIP/${EXTEN},5)
;exten => _1XXXX,n,NoOp(===== DIAL STATUS --> ${DIALSTATUS} =====)
;exten => _1XXXX,n,VoiceMail(${EXTEN}@default,u)
;exten => _1XXXX,n,Hangup()

; mailbox only
;exten => _1XXXX,1,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?3:2)
;exten => _1XXXX,2,Hangup()
;; mailbox exists, continue
;exten => _1XXXX,3,Ringing
;exten => _1XXXX,n,Wait(1)
;exten => _1XXXX,n,Answer
;exten => _1XXXX,n,Wait(1)
;exten => _1XXXX,n,VoiceMail(${EXTEN}@default,u)
;exten => _1XXXX,n,Hangup()

;; DIAL
; save dialed number (redial)
exten => _1XXXX,1,Set(DB(RepeatDial/${CALLERID(num)})=${EXTEN})

; unconditional forward
 exten => _1XXXX,n,Set(temp=${DB(CFIM/${CALLERID(num)})}) 
 exten => _1XXXX,n,GotoIf(${temp}?cfim:nocfim) 
 exten => _1XXXX,n(cfim),Dial(SIP/**${temp}@10.0.0.105)   ; Unconditional forward  
exten => _1XXXX,n,Goto(hup)
 exten => _1XXXX,n(nocfim),NoOp 

; dial number with delay
exten => _1XXXX,n,Dial(SIP/**${EXTEN}@10.0.0.105, 5)

; busy forward
 exten => _1XXXX,n,Set(temp=${DB(CFBS/${CALLERID(num)})}) 
 exten => _1XXXX,n,GotoIf(${temp}?cfbs:nocfbs) 
 exten => _1XXXX,n(cfbs),Dial(SIP/**${temp}@10.0.0.105) ; Forward on busy or unavailable  
exten => _1XXXX,n,Goto(hup)
; exten => _1XXXX,n(nocfbs),Busy
exten => _1XXXX,n(nocfbs),NoOp
; fallback to vmail instead of busy

; voicemail
exten => _1XXXX,n,NoOp(===== DIAL STATUS --> ${DIALSTATUS} =====)
exten => _1XXXX,n,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?vm:hup)
;; mailbox exists, continue
exten => _1XXXX,n(vm),Ringing
exten => _1XXXX,n,Wait(1)
exten => _1XXXX,n,Answer
exten => _1XXXX,n,Wait(1)
exten => _1XXXX,n,VoiceMail(${EXTEN}@default,u)
exten => _1XXXX,n(hup),Hangup()

Downloads

  • Asterisk, OpenSIPS, ODBC configuration tarball: sip.tar.bz2

Links

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138 Using TLS in OpenSIPS v2.2.x admin 2017-09-14 5104
137 opensips tls cfg admin 2017-09-14 5190
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 5729
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 2017-09-13 5057
134 OpenSIPS command line tricks admin 2017-09-13 5035
133 Fail2Ban Freeswitch How to secure admin 2017-09-12 5302
132 opensips.cfg. sample admin 2017-09-12 4993
131 Advanced SIP scenarios with Event-based-Routing admin 2017-09-11 5172
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 5644
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 2017-09-09 13988
128 rtpengine config basic and opensips configuration and command admin 2017-09-06 5338
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 2017-09-06 5161
126 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 5260
125 rtpengine install and config admin 2017-09-05 5245
124 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 5343
123 compile only the textops module make modules=modules/textops modules admin 2017-09-05 5219
122 opensips command /sbin/opensipsctl detail admin 2017-09-04 5314
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 2017-09-04 5248
120 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 5143
119 What is new in 2.3.0 opensips admin 2017-09-04 5967
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 2017-09-04 5508
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 5709
116 Using TLS in OpenSIPS v2.2.x configuration admin 2017-09-04 5395
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 2017-09-02 5364
114 You can install CDRTool in the following ways: admin 2017-09-01 5643
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 2017-09-01 5570
112 Opensips 2.32 download admin 2017-09-01 5339
111 OpenSIPS 2.3 install admin 2017-09-01 5678
110 JsSIP: The JavaScript SIP Library admin 2017-09-01 5608
109 WebSocket Transport using OpenSIPS admin 2017-09-01 5685
108 A2Billing and OpenSIPS – Part 1 admin 2017-08-29 5402
107 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 5312
106 A2Billing and OpenSIPS – Part 3 admin 2017-08-29 5527
105 OpenSIPS 2.3 philosophy admin 2017-08-17 6053
104 The timeline for OpenSIPS 2.3 is admin 2017-08-17 6184
103 OpenSIPS Control Panel and Homer integration admin 2017-08-17 6244
102 Opensips sip capture re designed admin 2017-07-16 5688
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 2015-04-04 10699
100 WebSocket Support in OpenSIPS 2.1 admin 2015-04-04 12062
99 OpenSIPS 2.1 (rc) is available, download now! admin 2015-03-22 10599
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 2015-02-25 12427
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 2014-11-02 19949
96 opensips.cfg for Asterisk admin 2014-10-20 22182
95 A2Billing and OpenSIPS config admin 2014-10-20 21511
94 Jitsi Videobridge meets WebRTC admin 2014-10-18 23187
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 2014-10-18 21134
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 2014-10-14 21325
91 Opensips TM module enables stateful processing of SIP transactions admin 2014-10-04 19036
90 kamailio.cfg configuration Example admin 2014-10-04 21298
89 opensips NAT Traversal Module admin 2014-10-02 20575
88 UAC Registrant Module admin 2014-09-28 22350
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 2014-08-24 21520
86 RTPPROXY Admin Guide admin 2014-08-24 21864
85 CANCEL MESSAGE not handled correctly admin 2014-08-23 21676
84 [Sipdroid] SIP data collection study tour admin 2014-08-23 22082
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 2014-08-23 22077
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 2014-08-23 21978
81 ICE: The ultimate way of beating NAT in SIP admin 2014-08-23 21624
80 Many OPENSIPS Configuration Examples This will Help you admin 2014-08-23 21280
79 Real-time Charging System for Telecom & ISP environments admin 2014-08-23 22032
78 OPENSIPS EBOOK admin 2014-08-21 22183
77 Opensips Documentation Function admin 2014-08-21 21869
76 Presence Tutorial OpenXCAP setup admin 2014-08-18 21470
75 Opensips Modules Documentation admin 2014-08-18 22142
74 A lightweight RPC library based on XML and HTTP admin 2014-08-18 21321
73 opensips Nat script with RTPPROXY - English Good perfect admin 2014-08-15 20127
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 2014-08-13 20342
71 Installation and configuration process record opensips opensips-cp admin 2014-08-13 46565
70 OpenSIPS as Homer Capture server admin 2014-08-13 19233
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 2014-08-13 21405
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 2014-08-12 21885
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 2014-08-12 21179
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 2014-08-11 22048
65 Kamailio Nat Traversal using RTPProxy admin 2014-08-11 21601
64 MediaProxy wiki page install configuration admin 2014-08-11 21659
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 40552
62 MediaProxy Installation Guide admin 2014-08-10 21172
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 2014-08-10 22410
60 Opensips Installation, How to. Good guide wiki page admin 2014-08-10 19435
59 OpenSIPS Installation Notes admin 2014-08-09 18945
58 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 32577
57 opensips 1.11.2 install Good Giide admin 2014-08-09 22509
56 fusionPBX install debian wheezy admin 2014-08-09 21370
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 2014-08-09 21676
54 SigIMS IMS Platform admin 2014-05-24 21909
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 2014-04-05 26389
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 24612
51 SIPSorcery admin 2014-03-18 22345
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 22772
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 47274
» SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 35695
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 30049
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 20950
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 22383
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 21604
43 Where to check OpenSIPS does not start? admin 2014-03-09 21716
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 22750
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 28525
40 Using the openSIPS Registrant Module admin 2014-03-09 23223
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 21190
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 22259
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 36114
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 20931
35 OpenSIPS Control Panel install guide admin 2014-03-06 22181
34 rtpproxy Module admin 2014-03-06 22039
33 MediaProxy Installation Guide admin 2014-03-06 30450
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 22925
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 22427
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 23446
29 Multimedia Service Platform admin 2014-03-06 21745
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 22528
27 Opensips Installation, How to. admin 2014-03-05 19226
26 100% CPU usage opensips admin 2014-03-05 21943
25 A2Billing and OpenSIPS admin 2014-03-04 23790
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 29252
23 Opensips install debian admin 2014-03-03 23061
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 23452
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 24664
20 My new toy: Bluebox-ng admin 2013-04-06 39056
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 41598
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 27547
17 The SIP Router Project admin 2013-04-06 26486
16 Kamailio :: A Quick Introduction admin 2013-04-06 24041
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 24331
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 29248
13 OpenSIPS vs Asterisk admin 2013-04-06 73013