chapter.voip.xml

<?xml version="1.0" encoding="UTF-8"?>
<!-- $Id: chapter.voip.xml 375 2012-03-21 03:00:42Z netkiller $ -->
<chapter id="index"><?dbhtml dir="voip" ?>
	<title>Voice over IP</title>
	<para>安装环境 ubuntu 7.10</para>
	<section id="gatekeeper">
		<title>Gnu Gatekeeper</title>
		<para>http://www.gnugk.org/</para>
	<section>
		<title>Gnu Gatekeeper Install</title>
		<screen>
sudo apt-get install gnugk
sudo apt-get install ohphone
		</screen>
		<para>start|stop|restart|force-reload</para>
		<screen>
netkiller@shenzhen:~$ sudo /etc/init.d/gnugk
Usage: /etc/init.d/gnugk {start|stop|restart|force-reload}
		</screen>
		<para>Start</para>
		<screen>
netkiller@shenzhen:~$ sudo /etc/init.d/gnugk start
Starting H.323 gatekeeper: gnugk.
netkiller@shenzhen:~$

netkiller@shenzhen:~$ sudo /etc/init.d/gnugk stop
Stopping H.323 gatekeeper: gnugk.
netkiller@shenzhen:~$
		</screen>
	</section>
	<section>
		<title>Gnu Gatekeeper Configure</title>
		<para>gatekeeper.ini</para>
		<screen>
[Gatekeeper::Main]
Fourtytwo=42
[GkStatus::Auth]
rule=allow
		</screen>
	</section>
	<section>
		<title>Gnu Gatekeeper Test</title>
		<para>How do I test Gatekeeper</para>
		<para>first, telnet tools</para>
		<screen>
netkiller@shenzhen:~$ telnet 127.0.0.1 7000
Trying 127.0.0.1...
Connected to 127.0.0.1.
Escape character is '^]'.
Version:
Gatekeeper(GNU) Version(2.2.5) Ext(pthreads=1,radius=1,mysql=1,pgsql=1,firebird=1,large_fdset=0,crypto/ssl=1) Build(Feb  2 2007, 21:39:07) Sys(Linux i686 2.6.20-15-server)
GkStatus: Version(2.0) Ext()
Toolkit: Version(1.0) Ext(basic)
Startup: Fri, 09 Nov 2007 17:26:23 -0500   Running: 0 days 00:08:34
;
		</screen>

		<section>
			<title>Part I - Microsoft Windows NetMeeting</title>
			<para>Windows XP</para>
			<para>Start NetMeeting</para>
			<para>Start->Run->conf</para>
			<screenshot>
				<graphic fileref="images/voip/step_1.png"/>
				<graphic fileref="images/voip/step_2.png"/>
				<graphic fileref="images/voip/step_3.png"/>
				<graphic fileref="images/voip/step_4.png"/>
				<graphic fileref="images/voip/step_5.png"/>
				<graphic fileref="images/voip/step_6.png"/>
				<graphic fileref="images/voip/step_7.png"/>
				<graphic fileref="images/voip/step_8.png"/>
				<graphic fileref="images/voip/step_9.png"/>
				<graphic fileref="images/voip/step_10.png"/>
				<graphic fileref="images/voip/firewall.png"/>
			</screenshot>
			<para>Tools -> Option -> Advence</para>
			<screenshot>
				<graphic fileref="images/voip/netmeeting-option.png"/>
			</screenshot>
			<para>网关守卫设置</para>
			<screenshot>
				<graphic fileref="images/voip/netmeeting_gateway.png"/>
			</screenshot>
		</section>
		<section>
			<title>Part II - ohphone</title>
			<para>For example:</para>
			<para>netkiller</para>
			<screen>
neo@machine1:~$ ohphone -l -a -u neo
			</screen>
			<para>neo</para>
			<screen>
netkiller@machine2:~$ ohphone -u netkiller neo
			</screen>

		</section>
	</section>
	</section>

	<section id="asterisk">
		<title>Asterisk (OpenSource Linux PBX that supports both SIP and H.323)</title>
		<para>http://www.asteriskpbx.com/</para>
		<screen>
netkiller@shenzhen:~$ apt-cache search Asterisk
asterisk-app-dtmftotext - Text entry application for Asterisk
asterisk-app-fax - Softfax application for Asterisk
asterisk-app-misdn-v110 - V.110 protocol handler for Asterisk
asterisk-chan-capi - Common ISDN API 2.0 implementation for Asterisk
asterisk-chan-misdn - mISDN support for Asterisk
asterisk-oh323 - oh323 channel driver for Asterisk
asterisk-prompt-de - German voice prompts for the Asterisk PBX
asterisk-prompt-es-co - Colombian Spanish voice prompts for Asterisk
asterisk-prompt-fr - French voice prompts for Asterisk
asterisk-prompt-it - Italian voice prompts for the Asterisk PBX
asterisk-prompt-se - Swedish voice prompts for Asterisk
asterisk-rate-engine - Asterisk least cost routing module
asterisk-sounds-extra - Additional sound files for the Asterisk PBX
destar - management interface for the Asterisk PBX
gastman - GUI tool for Asterisk administration and monitoring
iaxmodem - software modem with IAX2 connectivity
kiax - IAX VoIP softphone
libiax-dev - implementation of the Inter-Asterisk eXchange protocol (devel)
libiax0 - implementation of the Inter-Asterisk eXchange protocol
op-panel - switchboard type application for the Asterisk PBX
asterisk-prompt-es - Spanish prompts for the Asterisk PBX
asterisk - Open Source Private Branch Exchange (PBX)
asterisk-bristuff - Open Source Private Branch Exchange (PBX) - BRIstuff-enabled version
asterisk-classic - Open Source Private Branch Exchange (PBX) - original Digium version
asterisk-config - config files for asterisk
asterisk-dev - development files for asterisk
asterisk-doc - documentation for asterisk
asterisk-h323 - asterisk H.323 VoIP channel
asterisk-sounds-main - sound files for asterisk
asterisk-web-vmail - Web-based (CGI) voice mail interface for Asterisk
netkiller@shenzhen:~$
		</screen>
	</section>

	<section id="OpenSER">
		<title>OpenSER SIP Server</title>
		<para>http://www.openser.org/</para>
		<screen>
netkiller@shenzhen:~$ apt-cache search openser
openser - very fast and configurable SIP proxy
openser-cpl-module - CPL module (CPL interpreter engine) for OpenSER
openser-dbg - very fast and configurable SIP proxy [debug symbols]
openser-jabber-module - Jabber module (SIP-Jabber message translation) for OpenSER
openser-mysql-module - MySQL database connectivity module for OpenSER
openser-postgres-module - PostgreSQL database connectivity module for OpenSER
openser-radius-modules - radius modules for OpenSER
openser-unixodbc-module - unixODBC database connectivity module for OpenSER
		</screen>
	</section>
<!--
	<section>
		<title>Yate - Yet Another Telephony Engine (includes SIP to H.323 translation)</title>
		<para>http://yate.null.ro/pmwiki/</para>
	</section>
	<section>
		<title>VOCAL (includes a SIP to H.323 translator)</title>
		<para>http://www.vovida.org/</para>
	</section>
 -->

</chapter>