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://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

조회 수 :
33502
등록일 :
2014.03.12
12:32:50 (*.251.139.148)
엮인글 :
http://webs.co.kr/index.php?document_srl=39233&act=trackback&key=12e
게시글 주소 :
http://webs.co.kr/index.php?document_srl=39233
List of Articles
번호 제목 글쓴이 날짜 조회 수
162 Opensips Gateway between SIP and SMPP messages admin 2019-02-19 81
161 smpp sms opensips admin 2019-02-19 76
160 Busy Lamp Field (BLF) feature on Opensips 2.4.0 with Zoiper configuration admin 2018-05-29 1784
159 Documentation -> Tutorials -> WebSocket Transport using OpenSIPS admin 2018-05-17 1653
158 List of SIP response codes admin 2017-12-20 3309
157 opensips/modules/event_routing/ Push Notification Call pickup admin 2017-12-20 2870
156 opensips push notification How to detail file admin 2017-12-20 2775
155 OpenSIPS routing logic admin 2017-12-12 2851
154 OpenSIPS example configuration admin 2017-12-12 2829
153 opensips log output admin 2017-12-11 2837
152 opensips complete configuration example admin 2017-12-10 2924
151 Opensips1.6 ebook detail configuration and SIP signal and NAT etc file admin 2017-12-10 2921
150 dictionary.opensips radius admin 2017-12-09 3841
149 what is record_route() in opensips ? admin 2017-12-09 3765
148 what is loose_route() in opensips ? file admin 2017-12-09 3885
147 in opensips what is lookup(domain [, flags [, aor]]) admin 2017-12-09 3799
146 in opensips db_does_uri_exist() what is admin 2017-12-09 3645
145 in opensips what is has_totag() admin 2017-12-09 3795
144 opensips exec module admin 2017-12-08 3972
143 opensips push notification How to admin 2017-12-07 3747
142 OpenSIPS Module Interface admin 2017-12-07 3881
141 opensips configuration config explain easy basic 오픈쉽스 컨피그레이션 기본 설명 file admin 2017-12-07 3935
140 openssl 을 이용한 인증서 생성 절차를 정리한다. 개인키 CSR SSL 인증서 파일 생성 admin 2017-09-14 5010
139 Documentation -> Tutorials -> TLS opensips.cfg admin 2017-09-14 4790
138 Using TLS in OpenSIPS v2.2.x admin 2017-09-14 4775
137 opensips tls cfg admin 2017-09-14 4905
136 How to setup a Jabber / XMPP server on Debian 8 (jessie) using ejabberd admin 2017-09-13 5414
135 SIP to XMPP Gateway + SIP Presence Server opensips admin 2017-09-13 4753
134 OpenSIPS command line tricks admin 2017-09-13 4723
133 Fail2Ban Freeswitch How to secure admin 2017-09-12 5003
132 opensips.cfg. sample admin 2017-09-12 4713
131 Advanced SIP scenarios with Event-based-Routing admin 2017-09-11 4863
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 2017-09-11 5225
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 2017-09-09 12606
128 rtpengine config basic and opensips configuration and command admin 2017-09-06 5013
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 2017-09-06 4830
126 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 4965
125 rtpengine install and config admin 2017-09-05 4914
124 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 5006
123 compile only the textops module make modules=modules/textops modules admin 2017-09-05 4907
122 opensips command /sbin/opensipsctl detail admin 2017-09-04 4992
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 2017-09-04 4944
120 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 4822
119 What is new in 2.3.0 opensips admin 2017-09-04 5593
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 2017-09-04 5193
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 5397
116 Using TLS in OpenSIPS v2.2.x configuration admin 2017-09-04 5073
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 2017-09-02 5044
114 You can install CDRTool in the following ways: admin 2017-09-01 5261
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 2017-09-01 5256
112 Opensips 2.32 download admin 2017-09-01 5024
111 OpenSIPS 2.3 install admin 2017-09-01 5309
110 JsSIP: The JavaScript SIP Library admin 2017-09-01 5297
109 WebSocket Transport using OpenSIPS admin 2017-09-01 5379
108 A2Billing and OpenSIPS – Part 1 admin 2017-08-29 5105
107 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 5000
106 A2Billing and OpenSIPS – Part 3 admin 2017-08-29 5218
105 OpenSIPS 2.3 philosophy admin 2017-08-17 5714
104 The timeline for OpenSIPS 2.3 is admin 2017-08-17 5885
103 OpenSIPS Control Panel and Homer integration admin 2017-08-17 5827
102 Opensips sip capture re designed admin 2017-07-16 5383
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 2015-04-04 10358
100 WebSocket Support in OpenSIPS 2.1 admin 2015-04-04 11254
99 OpenSIPS 2.1 (rc) is available, download now! admin 2015-03-22 10250
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 2015-02-25 11964
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 2014-11-02 19573
96 opensips.cfg for Asterisk admin 2014-10-20 21770
95 A2Billing and OpenSIPS config admin 2014-10-20 21100
94 Jitsi Videobridge meets WebRTC admin 2014-10-18 22293
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 2014-10-18 20732
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 2014-10-14 20956
91 Opensips TM module enables stateful processing of SIP transactions admin 2014-10-04 18638
90 kamailio.cfg configuration Example admin 2014-10-04 20870
89 opensips NAT Traversal Module admin 2014-10-02 20189
88 UAC Registrant Module admin 2014-09-28 21938
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 2014-08-24 21081
86 RTPPROXY Admin Guide admin 2014-08-24 21462
85 CANCEL MESSAGE not handled correctly admin 2014-08-23 21226
84 [Sipdroid] SIP data collection study tour admin 2014-08-23 21678
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 2014-08-23 21686
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 2014-08-23 21582
81 ICE: The ultimate way of beating NAT in SIP admin 2014-08-23 21238
80 Many OPENSIPS Configuration Examples This will Help you admin 2014-08-23 20921
79 Real-time Charging System for Telecom & ISP environments admin 2014-08-23 21649
78 OPENSIPS EBOOK admin 2014-08-21 21771
77 Opensips Documentation Function admin 2014-08-21 21538
76 Presence Tutorial OpenXCAP setup admin 2014-08-18 21024
75 Opensips Modules Documentation admin 2014-08-18 21748
74 A lightweight RPC library based on XML and HTTP admin 2014-08-18 20961
73 opensips Nat script with RTPPROXY - English Good perfect admin 2014-08-15 19581
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 2014-08-13 19724
71 Installation and configuration process record opensips opensips-cp admin 2014-08-13 45659
70 OpenSIPS as Homer Capture server admin 2014-08-13 18872
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 2014-08-13 21012
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 2014-08-12 21485
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 2014-08-12 20732
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 2014-08-11 21654
65 Kamailio Nat Traversal using RTPProxy admin 2014-08-11 21237
64 MediaProxy wiki page install configuration admin 2014-08-11 21275
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 38509
62 MediaProxy Installation Guide admin 2014-08-10 20814
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 2014-08-10 22034
60 Opensips Installation, How to. Good guide wiki page admin 2014-08-10 18993
59 OpenSIPS Installation Notes admin 2014-08-09 18525
58 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 30840
57 opensips 1.11.2 install Good Giide admin 2014-08-09 21987
56 fusionPBX install debian wheezy admin 2014-08-09 21006
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 2014-08-09 21266
54 SigIMS IMS Platform admin 2014-05-24 21598
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 2014-04-05 25664
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 24171
51 SIPSorcery admin 2014-03-18 21990
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 22409
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 44039
» SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 33502
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 28827
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 20474
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 22076
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 21303
43 Where to check OpenSIPS does not start? admin 2014-03-09 21392
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 22412
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 27245
40 Using the openSIPS Registrant Module admin 2014-03-09 22839
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 20820
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 21952
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 34581
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 20530
35 OpenSIPS Control Panel install guide admin 2014-03-06 21721
34 rtpproxy Module admin 2014-03-06 21781
33 MediaProxy Installation Guide admin 2014-03-06 29227
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 22588
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 22098
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 23083
29 Multimedia Service Platform admin 2014-03-06 21426
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 22253
27 Opensips Installation, How to. admin 2014-03-05 18851
26 100% CPU usage opensips admin 2014-03-05 21622
25 A2Billing and OpenSIPS admin 2014-03-04 22877
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 28745
23 Opensips install debian admin 2014-03-03 22694
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 23139
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 24292
20 My new toy: Bluebox-ng admin 2013-04-06 38492
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 40090
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 27115
17 The SIP Router Project admin 2013-04-06 26126
16 Kamailio :: A Quick Introduction admin 2013-04-06 23531
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 23885
14 Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 28727
13 OpenSIPS vs Asterisk admin 2013-04-06 69698