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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

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127 WebSocket Transport using OpenSIPS configuration 웹 소켓 컨피그레이션 기본 admin 2017-09-06 5122
126 OpenSIPS basic configuration script 기본 컨피그 admin 2017-09-05 5224
125 rtpengine install and config admin 2017-09-05 5199
124 Installing RTPEngine on Ubuntu 14.04 admin 2017-09-05 5308
123 compile only the textops module make modules=modules/textops modules admin 2017-09-05 5178
122 opensips command /sbin/opensipsctl detail admin 2017-09-04 5277
121 2017 08 31 opensips 2.32 install debian8.8 module install compile err modules admin 2017-09-04 5204
120 Build-Depends debian 8.8 opensips 2.3 admin 2017-09-04 5097
119 What is new in 2.3.0 opensips admin 2017-09-04 5930
118 ubuntu 安装配置opensips,rtpproxy,mediaproxy admin 2017-09-04 5464
117 How to install Mediaproxy 2.5.2 on CentOS 6 64 bit admin 2017-09-04 5690
116 Using TLS in OpenSIPS v2.2.x configuration admin 2017-09-04 5358
115 How to 2.3 download , OpenSIPS new apt repository. DEBs for Debian / Ubuntu admin 2017-09-02 5317
114 You can install CDRTool in the following ways: admin 2017-09-01 5623
113 How to Install OpenSIPS 2.1.2 Server on Ubuntu 15.04 admin 2017-09-01 5527
112 Opensips 2.32 download admin 2017-09-01 5306
111 OpenSIPS 2.3 install admin 2017-09-01 5629
110 JsSIP: The JavaScript SIP Library admin 2017-09-01 5575
109 WebSocket Transport using OpenSIPS admin 2017-09-01 5665
108 A2Billing and OpenSIPS – Part 1 admin 2017-08-29 5367
107 A2Billing and OpenSIPS – Part 2 admin 2017-08-29 5274
106 A2Billing and OpenSIPS – Part 3 admin 2017-08-29 5488
105 OpenSIPS 2.3 philosophy admin 2017-08-17 6024
104 The timeline for OpenSIPS 2.3 is admin 2017-08-17 6161
103 OpenSIPS Control Panel and Homer integration admin 2017-08-17 6202
102 Opensips sip capture re designed admin 2017-07-16 5651
101 WebRTC with OpenSIPS WebSocket is a protocol provides full-duplex admin 2015-04-04 10673
100 WebSocket Support in OpenSIPS 2.1 admin 2015-04-04 11995
99 OpenSIPS 2.1 (rc) is available, download now! admin 2015-03-22 10558
98 Service Provision Using Asterisk & OpenSIPS - AstriCon 2014 admin 2015-02-25 12368
97 SIP Signaling-Messages OpenSIPS Running On Multicore Server file admin 2014-11-02 19897
96 opensips.cfg for Asterisk admin 2014-10-20 22139
95 A2Billing and OpenSIPS config admin 2014-10-20 21457
94 Jitsi Videobridge meets WebRTC admin 2014-10-18 23116
93 A Survey of Open Source Products for Building a SIP Communication Platform admin 2014-10-18 21079
92 Script Function , Module Index v1.11 함수 모듈 opensips admin 2014-10-14 21290
91 Opensips TM module enables stateful processing of SIP transactions admin 2014-10-04 18976
90 kamailio.cfg configuration Example admin 2014-10-04 21247
89 opensips NAT Traversal Module admin 2014-10-02 20535
88 UAC Registrant Module admin 2014-09-28 22302
87 MediaProxy 2.3.x & OpenSIPS 1.5.x Integration admin 2014-08-24 21472
86 RTPPROXY Admin Guide admin 2014-08-24 21823
85 CANCEL MESSAGE not handled correctly admin 2014-08-23 21632
84 [Sipdroid] SIP data collection study tour admin 2014-08-23 22042
83 [OpenSIPS-Users] Opensips 1.10 NAT radius aaa admin 2014-08-23 22036
82 OpenSIPS Consultancy Pricing module install Server 판매 또는 설치및 컨설팅 가이드 admin 2014-08-23 21939
81 ICE: The ultimate way of beating NAT in SIP admin 2014-08-23 21581
80 Many OPENSIPS Configuration Examples This will Help you admin 2014-08-23 21241
79 Real-time Charging System for Telecom & ISP environments admin 2014-08-23 21991
78 OPENSIPS EBOOK admin 2014-08-21 22138
77 Opensips Documentation Function admin 2014-08-21 21830
76 Presence Tutorial OpenXCAP setup admin 2014-08-18 21426
75 Opensips Modules Documentation admin 2014-08-18 22101
74 A lightweight RPC library based on XML and HTTP admin 2014-08-18 21274
73 opensips Nat script with RTPPROXY - English Good perfect admin 2014-08-15 20079
72 OpenSIPS Control Panel (OCP) Installation Guide Good admin 2014-08-13 20286
71 Installation and configuration process record opensips opensips-cp admin 2014-08-13 46508
70 OpenSIPS as Homer Capture server admin 2014-08-13 19188
69 OpenSIPS , default script , Types of Routs , Routing in SIP, Video lecture admin 2014-08-13 21364
68 Configuracion de Kamailio 3.3 con NAT Traversal y XCAP. admin 2014-08-12 21847
67 Under RHEL6.5 install OpenSIPS 1.11.1 tls admin 2014-08-12 21135
66 OpenSIPS/OpenSER-a versatile SIP Server cfg admin 2014-08-11 22007
65 Kamailio Nat Traversal using RTPProxy admin 2014-08-11 21563
64 MediaProxy wiki page install configuration admin 2014-08-11 21620
63 오픈소스 (사내)메신저 서버 구축, 오픈 파이어(openfire) 설치방법과 세팅 admin 2014-08-11 40414
62 MediaProxy Installation Guide admin 2014-08-10 21134
61 RTPProxy 1.2.x Installation & Integration with OpenSIPS 1.5x admin 2014-08-10 22372
60 Opensips Installation, How to. Good guide wiki page admin 2014-08-10 19384
59 OpenSIPS Installation Notes admin 2014-08-09 18899
58 Installation and configuration process record opensips 1.9.1 admin 2014-08-09 32451
57 opensips 1.11.2 install Good Giide admin 2014-08-09 22456
56 fusionPBX install debian wheezy admin 2014-08-09 21329
55 opensips 1.11.2 install guide good 인스톨 가이드 admin 2014-08-09 21626
54 SigIMS IMS Platform admin 2014-05-24 21878
53 2013 2012년 분야별 최고의 오픈소스 소프트웨어 124선 admin 2014-04-05 26333
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 24563
51 SIPSorcery admin 2014-03-18 22307
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 22727
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 47065
48 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 35547
47 Conference Support in Kamailio (OpenSER) admin 2014-03-12 29968
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 20903
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 22342
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 21579
43 Where to check OpenSIPS does not start? admin 2014-03-09 21673
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 22713
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 28416
40 Using the openSIPS Registrant Module admin 2014-03-09 23177
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 21136
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 22220
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 35997
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 20885
35 OpenSIPS Control Panel install guide admin 2014-03-06 22121
34 rtpproxy Module admin 2014-03-06 22008
33 MediaProxy Installation Guide admin 2014-03-06 30366
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 22891
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 22392
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 23399
29 Multimedia Service Platform admin 2014-03-06 21709
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 22489
27 Opensips Installation, How to. admin 2014-03-05 19182
26 100% CPU usage opensips admin 2014-03-05 21897
25 A2Billing and OpenSIPS admin 2014-03-04 23716
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 29215
23 Opensips install debian admin 2014-03-03 23025
22 Open Source VOIP applications, both clients and servers. admin 2013-11-20 23413
21 OfficeSIP Server is freeware VoIP, SIP server for Windows admin 2013-09-11 24621
20 My new toy: Bluebox-ng admin 2013-04-06 39010
19 Flooding Asterisk, Freeswitch and Kamailio with Metasploit admin 2013-04-06 41498
18 Asterisk Installation Asterisk Realtime configuration admin 2013-04-06 27505
17 The SIP Router Project admin 2013-04-06 26455
16 Kamailio :: A Quick Introduction admin 2013-04-06 23978
15 Welcome to the Smartvox Knowledgebase admin 2013-04-06 24289
» Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration using Asterisk Database admin 2013-04-06 29200
13 OpenSIPS vs Asterisk admin 2013-04-06 72782