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http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb

Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database


Author:
   Daniel-Constantin Mierla

This tutorial shows how to use Asterisk database to load the SIP user profile from within Kamailio configuration file. A typical use case is Kamailio as a SIP proxy router to scale Asterisk, by handling the user authentication and registration, letting one or a farm of Asterisks to deal with call handling (e.g., IVR, transconding, gatewaying, prepaid billing, a.s.o.).

The authentication module in Kamailio can be configured to connect to any database and fetch the password from custom table and column, therefore creation of a database view is not really required, unless you want for other purposes.

The document here presents the installation from sources, uses MySQL as database server and unixodbc for Asterisk realtime. The steps are given for Ubuntu/Debian operating systems.

Used versions are the latest stable releases from the both projects at the time of writing, respectively Kamailio v3.3.1 and Asterisk v10.7.0. To view what is new in Kamailio v3.3.xseries, visit the page:

Due to release policy of Kamailio project, where database structure and configuration file language are not changed in a stable branch, this tutorial will be valid for future releases numbered 3.3.x (e.g., 3.3.2, 3.3.3, …).



.

Previous release of this tutorial was using Kamailio 3.1.x series and Asterisk 1.6.2 and it is available at:

The improvements added to Kamailio configuration comparing with previous version:

  • no more need for extra table 'version' in Asterisk database
  • configuration option to handle short dialing
  • configuration option to drop 3XX redirect replies
  • optimized the NAT traversal part of configuration file
  • configuration option to enable voicemail redirection
  • restructuring of configuration file for simpler user authentication part

If you look for the other kind of integration approach (use of Kamailio database and create views to be accessed by Asterisk), follow next link:

Architecture

  • reuse as much as possible the default Asterisk relatime configuration
  • handle authentication in Kamailio
  • handle user location in Kamailio
  • routing of calls between local phones is managed by Asterisk
  • media services are handled by Asterisk according to Asterisk dialplan
  • routing of other SIP messages not related to calls are handled by Kamailio directly

Registration

Kamailio does authentication for registration. If successful, it notifies Asterisk with a new REGISTER that the phone is available at its IP.

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Call Initiation

Call authentication is handled by Kamailio. When a new calls arrives and it is authenticated, Kamailio forwards it to Asterisk. If destination number is online, Asterisk will send the call back to Kamailio since the contact of destination is Kamailio IP. Then Kamailio will do location lookup and send to destination phone IP.

img.php?width=0&height=0&antialias=1&edgesep=&round=1&shadow=1&scale=1&align=center&version=2010-11-24&md5=136f5563504c76bf925e6e0fc9a4bf3a

Requirements

Since many commands require root privileges, I assume you either know to use sudo to run the command or do su to root and run all commands as root:

sudo su -

MySQL Installation

MySQL server and client are included in all major Linux distributions. So is in Ubuntu which has version 5.5.x. To install the server and client, open a terminal and do:

apt-get install mysql-server

For a more detailed tutorial about MySQL installation on Ubuntu 12.04, see:

To install MySQL client library do:

apt-get install libmysqlclient-dev

Install UnixODBC

To install the UnixODBC devel libraries, run:

apt-get install unixodbc-dev

If your operating system does not provide a package for it, download the sources from http://www.unixodbc.org/, compile and install. Then tune the Asterisk compilation system if the unixodbc is not detected automatically.

To install the ODBC MySQL connector, run:

apt-get install libmyodbc

Asterisk Installation

Get Asterisk sources from http://www.asterisk.org. At this moment Asterisk 10.7.0 is the latest stable version.

cd /usr/local/src
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-10.7.0.tar.gz
tar xvfz asterisk-10.7.0.tar.gz
cd asterisk-10.7.0
./configure 

To enable ODBC storage for voicemail, run:

make menuselect

Then select option Voicemail Build Options, enable option ODBC_STORAGE. Save and exit.

Then compile and install:

make
make install

More notes about Asterisk installation process can be found in a dedicated chapter of Asterisk: The Definitive Guide book (although written for an older Asterisk version, it is still relevant for the version used in this tutorial).

Kamailio Installation

A step by step tutorial to install latest Kamailio 3.3.x from git is available at:

If you want to install from sources tarball:

cd /usr/local/src
wget http://www.kamailio.org/pub/kamailio/3.3.1/src/kamailio-3.3.1_src.tar.gz
tar xvfz kamailio-3.3.1_src.tar.gz
cd kamailio-3.3.1
make include_modules="db_mysql" cfg
make all
make install 

Kamailio Database

This database is required to store location records (phone contact addresses).

Use kamdbctl to create the database:

/usr/local/sbin/kamdbctl create

No other changes to Kamailio database structure are required. The SIP server will fetch the password from Asterisk database.

Asterisk Database

Execute next SQL script with mysql client:

CREATE DATABASE asterisk;
 
USE asterisk;
 
GRANT ALL ON asterisk.* TO asterisk@localhost IDENTIFIED BY 'asterisk_password';
 
CREATE TABLE `sipusers` (
  `id` int(11) NOT NULL AUTO_INCREMENT,
  `name` varchar(80) NOT NULL DEFAULT '',
  `host` varchar(31) NOT NULL DEFAULT '',
  `nat` varchar(5) NOT NULL DEFAULT 'no',
  `type` enum('user','peer','friend') NOT NULL DEFAULT 'friend',
  `accountcode` varchar(20) DEFAULT NULL,
  `amaflags` varchar(13) DEFAULT NULL,
  `call-limit` smallint(5) UNSIGNED DEFAULT NULL,
  `callgroup` varchar(10) DEFAULT NULL,
  `callerid` varchar(80) DEFAULT NULL,
  `cancallforward` char(3) DEFAULT 'yes',
  `canreinvite` char(3) DEFAULT 'yes',
  `context` varchar(80) DEFAULT NULL,
  `defaultip` varchar(15) DEFAULT NULL,
  `dtmfmode` varchar(7) DEFAULT NULL,
  `fromuser` varchar(80) DEFAULT NULL,
  `fromdomain` varchar(80) DEFAULT NULL,
  `insecure` varchar(4) DEFAULT NULL,
  `language` char(2) DEFAULT NULL,
  `mailbox` varchar(50) DEFAULT NULL,
  `md5secret` varchar(80) DEFAULT NULL,
  `deny` varchar(95) DEFAULT NULL,
  `permit` varchar(95) DEFAULT NULL,
  `mask` varchar(95) DEFAULT NULL,
  `musiconhold` varchar(100) DEFAULT NULL,
  `pickupgroup` varchar(10) DEFAULT NULL,
  `qualify` char(3) DEFAULT NULL,
  `regexten` varchar(80) DEFAULT NULL,
  `restrictcid` char(3) DEFAULT NULL,
  `rtptimeout` char(3) DEFAULT NULL,
  `rtpholdtimeout` char(3) DEFAULT NULL,
  `secret` varchar(80) DEFAULT NULL,
  `setvar` varchar(100) DEFAULT NULL,
  `disallow` varchar(100) DEFAULT NULL,
  `allow` varchar(100) DEFAULT NULL,
  `fullcontact` varchar(80) NOT NULL DEFAULT '',
  `ipaddr` varchar(45) DEFAULT NULL,
  `port` mediumint(5) UNSIGNED NOT NULL DEFAULT '0',
  `regserver` varchar(100) DEFAULT NULL,
  `regseconds` int(11) NOT NULL DEFAULT '0',
  `lastms` int(11) NOT NULL DEFAULT '0',
  `username` varchar(80) NOT NULL DEFAULT '',
  `defaultuser` varchar(80) NOT NULL DEFAULT '',
  `subscribecontext` varchar(80) DEFAULT NULL,
  `useragent` varchar(20) DEFAULT NULL,
  `sippasswd` varchar(80) DEFAULT NULL,
  PRIMARY KEY (`id`),
  UNIQUE KEY `name_uk` (`name`)
);
 
 
CREATE TABLE `sipregs` (
  `id` int(11) NOT NULL AUTO_INCREMENT,
  `name` varchar(80) NOT NULL DEFAULT '',
  `fullcontact` varchar(80) NOT NULL DEFAULT '',
  `ipaddr` varchar(45) DEFAULT NULL,
  `port` mediumint(5) UNSIGNED NOT NULL DEFAULT '0',
  `username` varchar(80) NOT NULL DEFAULT '',
  `regserver` varchar(100) DEFAULT NULL,
  `regseconds` int(11) NOT NULL DEFAULT '0',
  `defaultuser` varchar(80) NOT NULL DEFAULT '',
  `useragent` varchar(20) DEFAULT NULL,
  `lastms` int(11) DEFAULT NULL,
  PRIMARY KEY (`id`),
  UNIQUE KEY `name` (`name`)
);
 
 
CREATE TABLE `voiceboxes` (
  `uniqueid` int(4) NOT NULL AUTO_INCREMENT,
  `customer_id` varchar(10) DEFAULT NULL,
  `context` varchar(10) NOT NULL,
  `mailbox` varchar(10) NOT NULL,
  `password` varchar(12) NOT NULL,
  `fullname` varchar(150) DEFAULT NULL,
  `email` varchar(50) DEFAULT NULL,
  `pager` varchar(50) DEFAULT NULL,
  `tz` varchar(10) DEFAULT 'central',
  `attach` enum('yes','no') NOT NULL DEFAULT 'yes',
  `saycid` enum('yes','no') NOT NULL DEFAULT 'yes',
  `dialout` varchar(10) DEFAULT NULL,
  `callback` varchar(10) DEFAULT NULL,
  `review` enum('yes','no') NOT NULL DEFAULT 'no',
  `operator` enum('yes','no') NOT NULL DEFAULT 'no',
  `envelope` enum('yes','no') NOT NULL DEFAULT 'no',
  `sayduration` enum('yes','no') NOT NULL DEFAULT 'no',
  `saydurationm` tinyint(4) NOT NULL DEFAULT '1',
  `sendvoicemail` enum('yes','no') NOT NULL DEFAULT 'no',
  `delete` enum('yes','no') DEFAULT 'no',
  `nextaftercmd` enum('yes','no') NOT NULL DEFAULT 'yes',
  `forcename` enum('yes','no') NOT NULL DEFAULT 'no',
  `forcegreetings` enum('yes','no') NOT NULL DEFAULT 'no',
  `hidefromdir` enum('yes','no') NOT NULL DEFAULT 'yes',
  `stamp` timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP ON UPDATE CURRENT_TIMESTAMP,
  PRIMARY KEY (`uniqueid`),
  KEY `mailbox_context` (`mailbox`,`context`)
);
 
 
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,
  `flag` varchar(128) DEFAULT '',
  PRIMARY KEY (`id`),
  KEY `dir` (`dir`)
);

If you save it to asterisk.sql, then you can load it to MySQL server with:

mysql -u root -p <asterisk.sql

Before executing the SQL script, be sure you change the password for MySQL asterisk user, in line:

GRANT ALL ON asterisk.* to asterisk@localhost IDENTIFIED BY 'asterisk_password';

sipusers is the standard table required by Asterisk to store SIP user profile, with one extra column sippasswd where will be stored the password for SIP authentication. By default, Asterisk uses the column secret for SIP user password, but if that is filled in, Asterisk will ask for authentication again, resulting in double-authentication which we want to avoid.

sipregs is used to store SIP registrations. Registrations can be stored in sipusers tables as well, in case you do not want a separate table. Just omit the appropriate entry in /etc/asterisk/extconfig.conf.

voiceboxes is used to store voicemail box profiles and has the standard structure required by Asterisk. Storing voice box profiles in database allows to run several instances of Asterisk that can be load balanced or used in fail-over mode to store or listen to voice messages.

voicemessages is used to store voice messages and has the standard structure required by Asterisk. Storing voice messages in database allows to run several instances of Asterisk that can be load balanced or used in fail-over mode to store or listen to voice messages.

UnixODBC Configuration

Edit /etc/odbcinst.ini and add:

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

Edit /etc/odbc.ini and add:

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

Asterisk UnixODBC Configuration

Edit /etc/asterisk/res_odbc.conf and set:

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

Edit /etc/asterisk/extconfig.conf and set:

sipusers => odbc,asterisk,sipusers
sippeers => odbc,asterisk,sipusers
sipregs => odbc,asterisk,sipregs
voicemail => odbc,asterisk,voiceboxes

Asterisk Configuration

In case you need to cache the realtime users, then edit /etc/asterisk/sip.conf and set:

rtcachefriends=yes 

Be sure you update the listen IP and port as well if Asterisk is running on the same system with Kamailio.

Dialplan Configuration

It is up to you what dialplan you build in /etc/asterisk/extensions.conf. Practically is nothing special for this configuration, as phones will appear in Asterisk with contact address pointing to Kamailio IP and port.

Sample data

For testing purposes, here is a sample that can be plugged in /etc/asterisk/extensions.conf:

; our phones use 3 digit extensions, starting with 1
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup

It does the classic behaviour:

  • if phone is registered, route the call to it
  • if phone is unavailable, enter voicemail service
  • if phone is busy, enter voicemail service

In the Asterisk database, you can insert following records to create SIP users 101, 102 and 103:

INSERT INTO sipusers (name, username, host, sippasswd, fromuser, fromdomain, mailbox)
  VALUES ('101', '101', 'dynamic', '101', '101', 'yoursip.com', '101');
INSERT INTO sipusers (name, username, host, sippasswd, fromuser, fromdomain, mailbox)
  VALUES ('102', '102', 'dynamic', '102', '102', 'yoursip.com', '102');
INSERT INTO sipusers (name, username, host, sippasswd, fromuser, fromdomain, mailbox)
  VALUES ('103', '103', 'dynamic', '103', '103', 'yoursip.com', '103');
 
INSERT INTO sipregs(name) VALUES('101');
INSERT INTO sipregs(name) VALUES('102');
INSERT INTO sipregs(name) VALUES('103');
 
INSERT INTO voiceboxes(customer_id, context, mailbox, password) VALUES ('101', 'default', '101', '1234');
INSERT INTO voiceboxes(customer_id, context, mailbox, password) VALUES ('101', 'default', '102', '1234');
INSERT INTO voiceboxes(customer_id, context, mailbox, password) VALUES ('101', 'default', '103', '1234');

In case you use sipregs you have to create a record for each extension where to set the 'name' to value of 'name' from sipusers. The rest is populated by Asterisk from registrations.

Change the value of fromdomain (in the examples above yoursip.com) to your real SIP domain.

Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio.

Kamailio Configuration

This configuration file is an update of default Kamailio 3.1.x configuration file. It is easy to spot the changes with diff or following #!define WITH_ASTERISK (i.e., the parts within #!ifdef WITH_ASTERISK … #!endif.

Practically, if you want to disable the routing through Asterisk, remove the line:

#!define WITH_ASTERISK

The define directives are supported only starting with version 3.0.0. Also, registering to Asterisk in behalf of phones setting the contact address to Kamailio IP and port is a feature introduced in Kamailio 1.5.x, don't try this config with other forks of SER, working variants are Kamailio 3.0.x+ or SER v3.0.x+.

IP Addresses

Entire config file is pasted in the next sub-section. Do not forget to change the listen IP, port for Kamailio and Asterisk. In this example, Kamailio listens on IP 192.168.178.25 port 5060 and Asterisk listens on IP192.168.178.25 port 5080.

Also, if you created Asterisk or Kamailio databases with different names than specified above, or you changed the usernames and passwords to connect to MySQL server, do not forget to update DBURL and DBASTURLdefines.

Config File

Kamailio configuration file is located in /usr/local/etc/kamailio/kamailio.cfg when you install from sources or in /etc/kamailio/kamailio.cfg when you install from packages. Depending on your type of installation and CPU architecture, you may have to update the mpath config parameter to reflect the right folders where modules are installed.

#!KAMAILIO
 
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_ASTERISK
 
#
# Kamailio (OpenSER) SIP Server v3.3 - default configuration script
#     - web: http://www.kamailio.org
#     - git: http://sip-router.org
#
# Direct your questions about this file to: <sr-users@lists.sip-router.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode: 
#     - define WITH_DEBUG
#
# *** To enable mysql: 
#     - define WITH_MYSQL
#
# *** To enable authentication execute:
#     - enable mysql
#     - define WITH_AUTH
#     - add users using 'kamctl'
#
# *** To enable IP authentication execute:
#     - enable mysql
#     - enable authentication
#     - define WITH_IPAUTH
#     - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
#     - enable mysql
#     - define WITH_USRLOCDB
#
# *** To enable presence server execute:
#     - enable mysql
#     - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
#     - define WITH_NAT
#     - install RTPProxy: http://www.rtpproxy.org
#     - start RTPProxy:
#        rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enable PSTN gateway routing execute:
#     - define WITH_PSTN
#     - set the value of pstn.gw_ip
#     - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
#     - enable mysql
#     - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
#     - enable mysql
#     - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
#     - enable mysql
#     - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
#     - adjust CFGDIR/tls.cfg as needed
#     - define WITH_TLS
#
# *** To enable XMLRPC support execute:
#     - define WITH_XMLRPC
#     - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
#     - adjust pike and htable=>ipban settings as needed (default is
#       block if more than 16 requests in 2 seconds and ban for 300 seconds)
#     - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
#     - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
#     - define WITH_VOICEMAIL
#     - set the value of voicemail.srv_ip
#     - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
#     - enable mysql
#     - define WITH_ACCDB
#     - add following columns to database
#!ifdef ACCDB_COMMENT
  ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#!endif
 
####### Defined Values #########
 
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#       as: auth_db, acc, usrloc, a.s.o.
#!define DBURL "mysql://openser:openserrw@localhost/openser"
#!ifdef WITH_ASTERISK
#!define DBASTURL "mysql://asterisk:asterisk_password@localhost/asterisk"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
 
# - flags
#   FLT_ - per transaction (message) flags
#	FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
 
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
 
####### Global Parameters #########
 
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
 
memdbg=5
memlog=5
 
log_facility=LOG_LOCAL0
 
fork=yes
children=4
 
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
 
/* uncomment the next line to disable the auto discovery of local aliases
   based on reverse DNS on IPs (default on) */
#auto_aliases=no
 
/* add local domain aliases */
#alias="sip.mydomain.com"
 
/* uncomment and configure the following line if you want Kamailio to 
   bind on a specific interface/port/proto (default bind on all available) */
#listen=udp:10.0.0.10:5060
 
/* port to listen to
 * - can be specified more than once if needed to listen on many ports */
port=5060
 
#!ifdef WITH_TLS
enable_tls=yes
#!endif
 
# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605
 
####### Custom Parameters #########
 
# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#
 
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "PSTN GW Address"
#!endif
 
#!ifdef WITH_VOICEMAIL
# VoiceMail Routing on offline, busy or no answer
#
# - by default Voicemail server IP is empty to avoid misrouting
voicemail.srv_ip = "" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif
 
 
#!ifdef WITH_ASTERISK
asterisk.bindip = "192.168.178.25" desc "Asterisk IP Address"
asterisk.bindport = "5080" desc "Asterisk Port"
kamailio.bindip = "192.168.178.25" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif
 
####### Modules Section ########
 
# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules_k:modules"
#!else
mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"
#!endif
 
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
 
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
 
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
 
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
 
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
 
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
 
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
 
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
 
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
 
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
 
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
 
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
 
#!ifdef WITH_ASTERISK
loadmodule "uac.so"
#!endif
 
# ----------------- setting module-specific parameters ---------------
 
 
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
 
 
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
 
 
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
#!ifdef WITH_ASTERISK
modparam("rr", "append_fromtag", 1)
#!else
modparam("rr", "append_fromtag", 0)
#!endif
 
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 0)
 
 
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra", 
	"src_user=$fU;src_domain=$fd;src_ip=$si;"
	"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
	"src_user=$fU;src_domain=$fd;src_ip=$si;"
	"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif
 
 
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
 
 
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "load_credentials", "")
 
#!ifdef WITH_ASTERISK
modparam("auth_db", "user_column", "username")
modparam("auth_db", "password_column", "sippasswd")
modparam("auth_db", "db_url", DBASTURL)
modparam("auth_db", "version_table", 0)
#!else
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "use_domain", MULTIDOMAIN)
#!endif
 
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
 
#!endif
 
 
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
 
 
# ----- speedial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
 
 
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
 
 
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
 
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
 
 
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
 
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")
 
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
 
 
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")
#!endif
 
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
 
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
 
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
 
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
 
####### Routing Logic ########
 
 
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
 
	# per request initial checks
	route(REQINIT);
 
	# NAT detection
	route(NATDETECT);
 
	# handle requests within SIP dialogs
	route(WITHINDLG);
 
	### only initial requests (no To tag)
 
	# CANCEL processing
	if (is_method("CANCEL"))
	{
		if (t_check_trans())
			t_relay();
		exit;
	}
 
	t_check_trans();
 
	# authentication
	route(AUTH);
 
	# record routing for dialog forming requests (in case they are routed)
	# - remove preloaded route headers
	remove_hf("Route");
	if (is_method("INVITE|SUBSCRIBE"))
		record_route();
 
	# account only INVITEs
	if (is_method("INVITE"))
	{
		setflag(FLT_ACC); # do accounting
	}
 
	# dispatch requests to foreign domains
	route(SIPOUT);
 
	### requests for my local domains
 
	# handle presence related requests
	route(PRESENCE);
 
	# handle registrations
	route(REGISTRAR);
 
	if ($rU==$null)
	{
		# request with no Username in RURI
		sl_send_reply("484","Address Incomplete");
		exit;
	}
 
	# dispatch destinations to PSTN
	route(PSTN);
 
	# user location service
	route(LOCATION);
 
	route(RELAY);
}
 
 
route[RELAY] {
 
	# enable additional event routes for forwarded requests
	# - serial forking, RTP relaying handling, a.s.o.
	if (is_method("INVITE|SUBSCRIBE")) {
		t_on_branch("MANAGE_BRANCH");
		t_on_reply("MANAGE_REPLY");
	}
	if (is_method("INVITE")) {
		t_on_failure("MANAGE_FAILURE");
	}
 
	if (!t_relay()) {
		sl_reply_error();
	}
	exit;
}
 
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
	# flood dection from same IP and traffic ban for a while
	# be sure you exclude checking trusted peers, such as pstn gateways
	# - local host excluded (e.g., loop to self)
	if(src_ip!=myself)
	{
		if($sht(ipban=>$si)!=$null)
		{
			# ip is already blocked
			xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
			exit;
		}
		if (!pike_check_req())
		{
			xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
			$sht(ipban=>$si) = 1;
			exit;
		}
	}
#!endif
 
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}
 
	if(!sanity_check("1511", "7"))
	{
		xlog("Malformed SIP message from $si:$sp\n");
		exit;
	}
}
 
# Handle requests within SIP dialogs
route[WITHINDLG] {
	if (has_totag()) {
		# sequential request withing a dialog should
		# take the path determined by record-routing
		if (loose_route()) {
			if (is_method("BYE")) {
				setflag(FLT_ACC); # do accounting ...
				setflag(FLT_ACCFAILED); # ... even if the transaction fails
			}
			if ( is_method("ACK") ) {
				# ACK is forwarded statelessy
				route(NATMANAGE);
			}
			route(RELAY);
		} else {
			if (is_method("SUBSCRIBE") && uri == myself) {
				# in-dialog subscribe requests
				route(PRESENCE);
				exit;
			}
			if ( is_method("ACK") ) {
				if ( t_check_trans() ) {
					# no loose-route, but stateful ACK;
					# must be an ACK after a 487
					# or e.g. 404 from upstream server
					t_relay();
					exit;
				} else {
					# ACK without matching transaction ... ignore and discard
					exit;
				}
			}
			sl_send_reply("404","Not here");
		}
		exit;
	}
}
 
# Handle SIP registrations
route[REGISTRAR] {
	if (is_method("REGISTER"))
	{
		if(isflagset(FLT_NATS))
		{
			setbflag(FLB_NATB);
			# uncomment next line to do SIP NAT pinging 
			## setbflag(FLB_NATSIPPING);
		}
		if (!save("location"))
			sl_reply_error();
 
#!ifdef WITH_ASTERISK
		route(REGFWD);
#!endif
 
		exit;
	}
}
 
# USER location service
route[LOCATION] {
 
#!ifdef WITH_SPEEDIAL
	# search for short dialing - 2-digit extension
	if($rU=~"^[0-9][0-9]$")
		if(sd_lookup("speed_dial"))
			route(SIPOUT);
#!endif
 
#!ifdef WITH_ALIASDB
	# search in DB-based aliases
	if(alias_db_lookup("dbaliases"))
		route(SIPOUT);
#!endif
 
#!ifdef WITH_ASTERISK
	if(is_method("INVITE") && (!route(FROMASTERISK))) {
		# if new call from out there - send to Asterisk
		# - non-INVITE request are routed directly by Kamailio
		# - traffic from Asterisk is routed also directy by Kamailio
		route(TOASTERISK);
		exit;
	}
#!endif
 
	$avp(oexten) = $rU;
	if (!lookup("location")) {
		$var(rc) = $rc;
		route(TOVOICEMAIL);
		t_newtran();
		switch ($var(rc)) {
			case -1:
			case -3:
				send_reply("404", "Not Found");
				exit;
			case -2:
				send_reply("405", "Method Not Allowed");
				exit;
		}
	}
 
	# when routing via usrloc, log the missed calls also
	if (is_method("INVITE"))
	{
		setflag(FLT_ACCMISSED);
	}
}
 
# Presence server route
route[PRESENCE] {
	if(!is_method("PUBLISH|SUBSCRIBE"))
		return;
 
#!ifdef WITH_PRESENCE
	if (!t_newtran())
	{
		sl_reply_error();
		exit;
	};
 
	if(is_method("PUBLISH"))
	{
		handle_publish();
		t_release();
	}
	else
	if( is_method("SUBSCRIBE"))
	{
		handle_subscribe();
		t_release();
	}
	exit;
#!endif
 
	# if presence enabled, this part will not be executed
	if (is_method("PUBLISH") || $rU==$null)
	{
		sl_send_reply("404", "Not here");
		exit;
	}
	return;
}
 
# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
 
#!ifdef WITH_ASTERISK
	# do not auth traffic from Asterisk - trusted!
	if(route(FROMASTERISK))
		return;
#!endif
 
#!ifdef WITH_IPAUTH
	if((!is_method("REGISTER")) && allow_source_address())
	{
		# source IP allowed
		return;
	}
#!endif
 
	if (is_method("REGISTER") || from_uri==myself)
	{
		# authenticate requests
#!ifdef WITH_ASTERISK
		if (!auth_check("$fd", "sipusers", "1")) {
#!else
		if (!auth_check("$fd", "subscriber", "1")) {
#!endif
			auth_challenge("$fd", "0");
			exit;
		}
		# user authenticated - remove auth header
		if(!is_method("REGISTER|PUBLISH"))
			consume_credentials();
	}
	# if caller is not local subscriber, then check if it calls
	# a local destination, otherwise deny, not an open relay here
	if (from_uri!=myself && uri!=myself)
	{
		sl_send_reply("403","Not relaying");
		exit;
	}
 
#!endif
	return;
}
 
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
	force_rport();
	if (nat_uac_test("19")) {
		if (is_method("REGISTER")) {
			fix_nated_register();
		} else {
			fix_nated_contact();
		}
		setflag(FLT_NATS);
	}
#!endif
	return;
}
 
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
	if (is_request()) {
		if(has_totag()) {
			if(check_route_param("nat=yes")) {
				setbflag(FLB_NATB);
			}
		}
	}
	if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
		return;
 
	rtpproxy_manage();
 
	if (is_request()) {
		if (!has_totag()) {
			add_rr_param(";nat=yes");
		}
	}
	if (is_reply()) {
		if(isbflagset(FLB_NATB)) {
			fix_nated_contact();
		}
	}
#!endif
	return;
}
 
# Routing to foreign domains
route[SIPOUT] {
	if (!uri==myself)
	{
		append_hf("P-hint: outbound\r\n");
		route(RELAY);
	}
}
 
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
	# check if PSTN GW IP is defined
	if (strempty($sel(cfg_get.pstn.gw_ip))) {
		xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
		return;
	}
 
	# route to PSTN dialed numbers starting with '+' or '00'
	#     (international format)
	# - update the condition to match your dialing rules for PSTN routing
	if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
		return;
 
	# only local users allowed to call
	if(from_uri!=myself) {
		sl_send_reply("403", "Not Allowed");
		exit;
	}
 
	$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
 
	route(RELAY);
	exit;
#!endif
 
	return;
}
 
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
	# allow XMLRPC from localhost
	if ((method=="POST" || method=="GET")
			&& (src_ip==127.0.0.1)) {
		# close connection only for xmlrpclib user agents (there is a bug in
		# xmlrpclib: it waits for EOF before interpreting the response).
		if ($hdr(User-Agent) =~ "xmlrpclib")
			set_reply_close();
		set_reply_no_connect();
		dispatch_rpc();
		exit;
	}
	send_reply("403", "Forbidden");
	exit;
}
#!endif
 
# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
	if(!is_method("INVITE"))
		return;
 
	# check if VoiceMail server IP is defined
	if (strempty($sel(cfg_get.voicemail.srv_ip))) {
		xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
		return;
	}
	if($avp(oexten)==$null)
		return;
 
	$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
				+ ":" + $sel(cfg_get.voicemail.srv_port);
	route(RELAY);
	exit;
#!endif
 
	return;
}
 
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
	xdbg("new branch [$T_branch_idx] to $ru\n");
	route(NATMANAGE);
}
 
# manage incoming replies
onreply_route[MANAGE_REPLY] {
	xdbg("incoming reply\n");
	if(status=~"[12][0-9][0-9]")
		route(NATMANAGE);
}
 
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
	route(NATMANAGE);
 
	if (t_is_canceled()) {
		exit;
	}
 
#!ifdef WITH_BLOCK3XX
	# block call redirect based on 3xx replies.
	if (t_check_status("3[0-9][0-9]")) {
		t_reply("404","Not found");
		exit;
	}
#!endif
 
#!ifdef WITH_VOICEMAIL
	# serial forking
	# - route to voicemail on busy or no answer (timeout)
	if (t_check_status("486|408")) {
		route(TOVOICEMAIL);
		exit;
	}
#!endif
}
 
#!ifdef WITH_ASTERISK
# Test if coming from Asterisk
route[FROMASTERISK] {
	if($si==$sel(cfg_get.asterisk.bindip)
			&& $sp==$sel(cfg_get.asterisk.bindport))
		return 1;
	return -1;
}
 
# Send to Asterisk
route[TOASTERISK] {
	$du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
			+ $sel(cfg_get.asterisk.bindport);
	route(RELAY);
	exit;
}
 
# Forward REGISTER to Asterisk
route[REGFWD] {
	if(!is_method("REGISTER"))
	{
		return;
	}
	$var(rip) = $sel(cfg_get.asterisk.bindip);
	$uac_req(method)="REGISTER";
	$uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport);
	$uac_req(furi)="sip:" + $au + "@" + $var(rip);
	$uac_req(turi)="sip:" + $au + "@" + $var(rip);
	$uac_req(hdrs)="Contact: <sip:" + $au + "@"
				+ $sel(cfg_get.kamailio.bindip)
				+ ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
	if($sel(contact.expires) != $null)
		$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n";
	else
		$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n";
	uac_req_send();
}
#!endif

Config Remarks

  • REGISTER request sent to Asterisk is triggered by a REGISTER coming from phone, but is built from scratch and sent with uac_req_send().
  • IP and pot for kamailio set via custom global parameters kamailio.bindip and kamailio.bindport are used to build the contact for REGISTER request sent to Asterisk
  • any INVITE received from phones (not coming from Asterisk) is authenticated and then sent to Asterisk
  • Asterisk will automatically send back to Kamailio the INVITEs for online SIP phones (as the contact in Asterisk sipregs points to Kamailio IP and port)
  • any INVITE received from Asterisk is handled via user location and then sent to destination phone

Other Benefits

With such architecture, several other benefits can be achieved quickly:

  • increase of security - Kamailio handling SIP signaling only, can absorb easier the flooding attacks, protecting Asterisk
  • transport layer gatewaying - Kamailio has mature implementations for UDP, TCP, TLS and SCTP, therefore you can use it in front of Asterisk to translate between these protocols
  • load balancing - you can use several instances of Asterisk, Kamailio can do load balancing among them
  • high availability - Kamailio can be configured to re-route the call if selected Asterisk box does not react in a given period of time, e.g., if one Asterisk is not responsive in 2 sec, sent the call to another Asterisk

See also

조회 수 :
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131 Advanced SIP scenarios with Event-based-Routing admin 4844   2017-09-11
 
130 PUSH SERVER 푸시서버 안드로이드 애플 admin 5192   2017-09-11
 
129 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅(리눅스 기준) admin 12533   2017-09-09
 
128 rtpengine config basic and opensips configuration and command admin 4987   2017-09-06
 
127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 4806   2017-09-06
 
126 OpenSIPS basic configuration script 기본 컨피그 admin 4943   2017-09-05
 
125 rtpengine install and config admin 4892   2017-09-05
 
124 Installing RTPEngine on Ubuntu 14.04 admin 4991   2017-09-05
 
123 compile only the textops module make modules=modules/textops modules admin 4878   2017-09-05
 
122 opensips command /sbin/opensipsctl detail admin 4971   2017-09-04
 
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 4926   2017-09-04
 
120 Build-Depends debian 8.8 opensips 2.3 admin 4801   2017-09-04
 
119 What is new in 2.3.0 opensips admin 5563   2017-09-04
 
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 5174   2017-09-04
 
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 5361   2017-09-04
 
116 Using TLS in OpenSIPS v2.2.x configuration admin 5045   2017-09-04
 
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 5018   2017-09-02
 
114 You can install CDRTool in the following ways: admin 5236   2017-09-01
 
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 5234   2017-09-01
 
112 Opensips 2.32 download admin 5004   2017-09-01
 
111 OpenSIPS 2.3 install admin 5278   2017-09-01
 
110 JsSIP: The JavaScript SIP Library admin 5272   2017-09-01
 
109 WebSocket Transport using OpenSIPS admin 5338   2017-09-01
 
108 A2Billing and OpenSIPS – Part 1 admin 5084   2017-08-29
 
107 A2Billing and OpenSIPS – Part 2 admin 4980   2017-08-29
 
106 A2Billing and OpenSIPS – Part 3 admin 5198   2017-08-29
 
105 OpenSIPS 2.3 philosophy admin 5683   2017-08-17
 
104 The timeline for OpenSIPS 2.3 is admin 5855   2017-08-17
 
103 OpenSIPS Control Panel and Homer integration admin 5791   2017-08-17
 
102 Opensips sip capture re designed admin 5368   2017-07-16
 
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 10330   2015-04-04
 
100 WebSocket Support in OpenSIPS 2.1 admin 11207   2015-04-04
 
99 OpenSIPS 2.1 (rc) is available, download now! admin 10224   2015-03-22
 
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 11943   2015-02-25
 
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 19552   2014-11-02
 
96 opensips.cfg for Asterisk admin 21741   2014-10-20
 
95 A2Billing and OpenSIPS config admin 21076   2014-10-20
 
94 Jitsi Videobridge meets WebRTC admin 22238   2014-10-18
 
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 20701   2014-10-18
 
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 20934   2014-10-14
 
91 Opensips TM module enables stateful processing of SIP transactions admin 18616   2014-10-04
 
90 kamailio.cfg configuration Example admin 20841   2014-10-04
 
89 opensips NAT Traversal Module admin 20155   2014-10-02
 
88 UAC Registrant Module admin 21915   2014-09-28
 
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 21056   2014-08-24
 
86 RTPPROXY Admin Guide admin 21436   2014-08-24
 
85 CANCEL MESSAGE not handled correctly admin 21201   2014-08-23
 
84 [Sipdroid] SIP data collection study tour admin 21649   2014-08-23
 
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 21658   2014-08-23
 
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 21560   2014-08-23
 
81 ICE: The ultimate way of beating NAT in SIP admin 21210   2014-08-23
 
80 Many OPENSIPS Configuration Examples This will Help you admin 20897   2014-08-23
 
79 Real-time Charging System for Telecom & ISP environments admin 21617   2014-08-23
 
78 OPENSIPS EBOOK admin 21744   2014-08-21
 
77 Opensips Documentation Function admin 21519   2014-08-21
 
76 Presence Tutorial OpenXCAP setup admin 20994   2014-08-18
 
75 Opensips Modules Documentation admin 21717   2014-08-18
 
74 A lightweight RPC library based on XML and HTTP admin 20937   2014-08-18
 
73 opensips Nat script with RTPPROXY - English Good perfect admin 19560   2014-08-15
 
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 19682   2014-08-13
 
71 Installation and configuration process record opensips opensips-cp admin 45565   2014-08-13
 
70 OpenSIPS as Homer Capture server admin 18847   2014-08-13
 
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 20990   2014-08-13
 
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 21462   2014-08-12
 
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 20704   2014-08-12
 
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 21627   2014-08-11
 
65 Kamailio Nat Traversal using RTPProxy admin 21210   2014-08-11
 
64 MediaProxy wiki page install configuration admin 21250   2014-08-11
 
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 38409   2014-08-11
 
62 MediaProxy Installation Guide admin 20794   2014-08-10
 
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 22007   2014-08-10
 
60 Opensips Installation, How to. Good guide wiki page admin 18960   2014-08-10
 
59 OpenSIPS Installation Notes admin 18497   2014-08-09
 
58 Installation and configuration process record opensips 1.9.1 admin 30751   2014-08-09
 
57 opensips 1.11.2 install Good Giide admin 21959   2014-08-09
 
56 fusionPBX install debian wheezy admin 20975   2014-08-09
 
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 21231   2014-08-09
 
54 SigIMS IMS Platform admin 21570   2014-05-24
 
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 25616   2014-04-05
 
52 Video conference server OpenMCU-ru - Introduction admin 24139   2014-04-01
 
51 SIPSorcery admin 21962   2014-03-18
 
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 22382   2014-03-12
 
49 telepresence: Open Source SIP Telepresence/MCU admin 43890   2014-03-12
 
48 SIP PBX - OpenSIPS and Asterisk configuration admin 33401   2014-03-12
 
47 Conference Support in Kamailio (OpenSER) admin 28767   2014-03-12
 
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 20446   2014-03-12
 
45 The Impact of TLS on SIP Server Performance file admin 22047   2014-03-12
 
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 21281   2014-03-12
 
43 Where to check OpenSIPS does not start? admin 21372   2014-03-09
 
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 22388   2014-03-09
 
41 Kamailo OpenSIPs installation on Debian admin 27189   2014-03-09
 
40 Using the openSIPS Registrant Module admin 22816   2014-03-09
 
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 20789   2014-03-07
 
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 21927   2014-03-07
 
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 34501   2014-03-07
 
36 OpenSIPS Control Panel (OCP) Installation Guide admin 20498   2014-03-06
 
35 OpenSIPS Control Panel install guide admin 21679   2014-03-06
 
34 rtpproxy Module admin 21758   2014-03-06
 
33 MediaProxy Installation Guide admin 29164   2014-03-06
 
32 How to install OpenSIPS on CentOS debian module add xcap admin 22562   2014-03-06
 
31 Problem with presence_xml module Opensips 1.9 admin 22076   2014-03-06
 
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 23049   2014-03-06
 
29 Multimedia Service Platform admin 21406   2014-03-06
 
28 How to install OpenSIPS on CentOS Debian etc admin 22236   2014-03-05
 
27 Opensips Installation, How to. admin 18822   2014-03-05
 
26 100% CPU usage opensips admin 21604   2014-03-05
 
25 A2Billing and OpenSIPS admin 22808   2014-03-04
 
24 Opensips_1.9 install guide this is great I like this admin 28702   2014-03-04
 
23 Opensips install debian admin 22667   2014-03-03
 
22 Open Source VOIP applications, both clients and servers. admin 23114   2013-11-20
 
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 24264   2013-09-11
 
20 My new toy: Bluebox-ng admin 38455   2013-04-06
 
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 40016   2013-04-06
 
18 Asterisk Installation Asterisk Realtime configuration admin 27087   2013-04-06
 
17 The SIP Router Project admin 26090   2013-04-06
 
16 Kamailio :: A Quick Introduction admin 23497   2013-04-06
 
15 Welcome to the Smartvox Knowledgebase admin 23862   2013-04-06
 
» Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 28702   2013-04-06
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdbKamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database Author: Daniel-Constantin Mierla This tutorial shows how to use A...  
13 OpenSIPS vs Asterisk admin 69559   2013-04-06