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Revision as of 21:19, 4 December 2010

Advanced Topics - Setting Up a VOIP Server
Prev Bering-uClibc 4.x - User Guide Next


Yate

This material copied directly from http://leaf.sourceforge.net/doc/bucu-voip.html - needs to be checked/updated for Bering-uClibc 4.x!
Davidmbrooke 20:46, 16 November 2010 (UTC)

Introduction

This Howto describes a Voice-over-IP (VoIP) solution based on Yate. Yate is a small but powerful IP telephony engine.

Requirements

To install Yate on Bering-uClibc based system the following packages are required:

yate.lrp
libcxx.lrp
libm.lrp
lpthread.lrp

Configuration

To allow SIP clients to register set registrar=yes in ysipchan.conf. You have to set the username and passwords for those clients in regfile.conf like this:

[someusername]
password=something

The regexroute module provides a simple way of routing telephony calls inside Yate. This module describes the routes using a configuration file in which each number is matched using regular expressions. The config file is named regexroute.conf and contains numerous examples.

The accfile module allows Yate to act as a SIP or AIX client, this can be used to connect Yate to an other VOIP server or provider.

[test_sip]
 enabled=yes
 protocol=sip
 username=me
 description=Test SIP account
 ;interval=600
 formats=alaw,mulaw
 password=1234
 ;number=1234
 ;domain=somewhere.org
 registrar=10.0.0.1:5060
 ;outbound=10.0.0.1:5061
 ;localaddress=auto

An example on how to connect Yate to FWD (Free World Dialup) can be found in yate's wiki.

Firewall settings

In most situations you don't install a firewall VoIP server itself. If you install a VoIP server behind a firewall you need to open a SIP and a RTP range.

SIP uses udp port 5060, RTP (udp) ports are not hard defined but depends very much upon the application. In Yate the range of ports can be defined in yrtpchan.conf.


Asterisk

TODO - Volunteers?
Need to fix the code first! See https://sourceforge.net/apps/trac/leaf/ticket/5



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