Beginners in Yate
(→Test the setup)
|Line 85:||Line 85:|
The file used is [[SIP Configuration File|ysipchan.conf]]. No configuration is needed in this file because by default Yate will bind with all the network interfaces on your server on port 5060. If other programs use this port then you have to use another free port and put it in [general] section.
Revision as of 14:49, 10 March 2015
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.
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.
Note The instructions below are suitable for linux platforms. For other platforms go to Download page and follow the instructions from there. After you install Yate you can skip chapter How to get Yate source from SVN and go to chapter Configuration Files.
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:
- A subversion (svn) client
As root 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:
How to run
- to run in debug mode:
- to run in the daemon mode:
For more details about what parameters that can be set when Yate starts you can give command
As an advice for debbuging purpose run Yate with this parameters:
-v Verbose debugging (you can use more than once) -d Daemonify, suppress output unless logged -l filename Log to file -Dt Timestamp debugging messages relative to program start
The files that you can configure are in /usr/src/yate-SVN/conf.d.
Note Each file has a .sample termination. You have to create a new file with the same name but with .conf.
Each parameter in the files have a section that is in brackets. Comment in this files are done by using ;.
Go to /usr/src/yate-SVN/conf.d where Yate was installed and rename regfile.conf.sample into regfile.conf.
Then edit regfile.conf to add users.
We are going to add 2 users:user 100 with password 001 and user 200 with password 002 like this:
 password=001  password=002
There is another way of adding users if you wish to use a database, the file to use is register.conf.
The file used is ysipchan.conf. No configuration is needed in this file because by default Yate will bind with all the network interfaces on your server on port 5060. If other programs use this port then you have to use another free port and put it in [general] section.
There is no need to define any routing for registered SIP users on the machine. Yate will know to route calls between the users defined in regfile.conf.
To add authentication requirement for all inbound calls add in the regexroute.conf file:
To define routing to other registered users, PSTN, gateways you need to edit regexroute.conf.
To register users in a database you can use register.conf. And then make your route rules in regexroute.conf.
You have to configure the users from regfile on two SIP phone (you could use a SIP softphone as well) to call Yate Server.
Test the setup
If Yate was running when you wrote in configuration file, 'reload Yate' so that the changes will take place.
From the phone, dial one of the following:
99991001 - you should hear dial tone 99991002 - you should hear busy tone 99991003 - you should hear ring tone
These are numbers defined for testing purposes and you can find all of them in the "regexroute.conf" file. When you call one of those numbers, you'll get a standard telephony tone.
Then make a call from one phone to the other and check the audio, if you can hear the other end then you may have a NAT problem.
Also to see the flow of the messages, use a telnet client. Connect like this:
telnet localhost 5038
And write next commands to enable the debug and color the output:
debug on color on
Then make the call and watch SIP messages flow.