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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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50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 23579
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 55351
» SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 41634
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 33056
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 22021
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 22957
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 22192
43 Where to check OpenSIPS does not start? admin 2014-03-09 22393