Beginners in Yate

Yate (acronym for Yet Another Telephony Engine) is a next-generation telephony engine, is a free and open source communications software with support for video, voice and instant messaging.

Based on Voice over Internet Protocol (VoIP) and PSTN, it can easily be extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, ISDN PRI, BRI, and SS7.

YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.

It is written in C++, having in mind a modular design, allowing the use of scripting languages like Perl, Python or PHP to create external functionalities.

How to get Yate source from SVN

Since you are going the full process of fetching and building Yate you will need the following:

• Basic software development tools:
  • The [make] program (it won't build with the BSD make)
  • The C++ compiler of the Gnu suite ([[1]]) and its libraries
  • The [[2]] configuration script builder
• A [[3]] (svn) client

Go to /usr/src or where ever you'd like to store source code. Once you have the svn client installed getting the sources is a simple command:

svn checkout http://voip.null.ro/svn/yate/trunk yate-SVN
cd yate-SVN

First command will fetch a copy of the SVN TRUNK (where the code is committed) in a new directory called yate-SVN. The second command will change your current directory to the Yate sources directory.

For more information go to page Installation.

How to compile

To generate configure file run this, then configure the source code:

./autogen.sh
./configure

compile it:

make

How to run

• to run in debug mode:

./run -vvvvvv

• to run in the daemon mode:

./run -d

Adding Users

You'll have to edit regfile.conf to add users. If you want to add user 100 with password 001, you need to add this:

[100]
password=001

SIP Configuration

'ysipchan.conf' has two sections. In [general] section you can define the listening port and IP address to bind to. Also you can enable/disable user registration.

Section [codecs] is used to configure codecs:

[codecs]

; This section allows to individually enable or disable 
the codecs

default=off
mulaw=yes
alaw=yes

Routing

To define routing to other registered users, PSTN, gateways we need to edit 'regexroute.conf'.

There is no need to define any routing for registered SIP users on the same machine.