Difference between revisions of "Bering-uClibc 4.x - User Guide - Advanced Topics - Setting Up a VOIP Server"

From bering-uClibc
Jump to: navigation, search
(Asterisk: Updated for version loaded into Git, for release as part of Bering-uClibc 4.1)
(A Simple Test Configuration: Expanded)
Line 124: Line 124:
 
However, a very simple but working configuration can be achieved with the following steps:
 
However, a very simple but working configuration can be achieved with the following steps:
 
# Install the <code class="filename">asterisk.lrp</code> and <code class="filename">astsmpls.lrp</code> Packages along with any missing pre-requisites.
 
# Install the <code class="filename">asterisk.lrp</code> and <code class="filename">astsmpls.lrp</code> Packages along with any missing pre-requisites.
# Copy <code class="filename">/etc/asterisk/samples/asterisk.conf</code> to <code class="filename">/etc/asterisk/asterisk.conf</code>
+
# Copy <code class="filename">/etc/asterisk/samples/*.conf</code> to <code class="filename">/etc/asterisk/</code>
To be continued...
+
# Edit <code class="filename">/etc/asterisk/asterisk.conf</code> and:
 +
#* Remove the comment character at the start of the line for <code>highpriority = yes</code>
 +
# Edit <code class="filename">/etc/asterisk/sip.conf</code> and:
 +
#* Remove the comment character at the start of the line for <code>match_auth_username=yes</code>
 +
#* Remove the comment character at the start of the line for <code>allowguest=no</code>
 +
#* Add an entry for a "peer" (SIP phone) - something like the following:<pre>[host-mac_address_without_colons]&#10;type=friend&#10;secret=<your password here>&#10;deny=0.0.0.0/0&#10;permit=192.168.1.0/24&#10;qualify=100&#10;nat=no&#10;host=dynamic&#10;directmedia=yes</pre>
 +
# Ensure that any firewall running on the VOIP server is configured to permit UDP connections on port 5060
 +
# Start the asterisk process in verbose mode in the foreground by running the following command: <pre>asterisk -vvvc</pre>
 +
'''Incomplete''' but to be continued... [[User:Davidmbrooke|Davidmbrooke]] 20:09, 22 October 2011 (UTC)
  
 
===Further Reading===
 
===Further Reading===

Revision as of 20:09, 22 October 2011

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


Yate

Introduction

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

The yate.lrp Package in Bering-uClibc 4.x is based on Yate version 3. This is much more advanced than Yate version 1 which was used for Bering-uClibc 3.x and includes many more Yate modules, each with their own configuration file. Since many of these Yate modules will only be of interest to advanced users, the Bering-uClibc 4.x yate.lrp Package includes a large number of Yate modules (and their configuration files) but only loads a small number of Yate modules by default (the same ones as were available for Yate version 1 on Bering-uClibc 3.x) and only lists the configuration files for these default Yate modules in the lrcfg menu.

Requirements

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

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

Yate Basics

Startup and Shutdown

The Yate processes are stopped and started via init script /etc/init.d/yate. This is executed automatically at boot time, or the Yate processes can be started manually with the following command:

svi yate start

Logging

Log output is sent to /var/log/yate.log By default the logging is fairly minimal. Extra log output can be configured by adjusting the [debug] settings in /etc/yate/yate.conf or by using the Remote Manager.

Test Numbers

Yate comes pre-configured with some test numbers, which are specified at the end of /etc/yate/regexroute.conf:

; The following are for testing purposes                                        
^99991001$=tone/dial                                                            
^99991002$=tone/busy                                                            
^99991003$=tone/ring                                                            
^99991004$=tone/specdial                                                        
^99991005$=tone/congestion                                                      
^99991006$=tone/outoforder                                                      
^99991007$=tone/milliwatt 

For example, after dialling 99991003@192.168.1.254 (assuming that is the IPv4 address of your Yate server) you should hear a (US) ring tone.

Configuration Recipes

Yate has a number of configuration files in directory /etc/yate/. Only some of those are listed in the Yate section of the LEAF configuration menu.

SIP Client Registration

To allow SIP clients to register edit file ysipchan.conf and un-comment the registrar entry.

; registrar: bool: Allow the SIP module to receive registration requests        
registrar=enable 

You have to set the username and passwords for those clients in regfile.conf like this:

[someusername]
password=something

Call Routing Configuration

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.

Accounts

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

Status

As of June 2011 Asterisk version 1.8.4.3 (the latest upstream version) has been ported to Bering-uClibc 4.x and will be released as part of Bering-uClibc 4.1 (initially via 4.1-beta1). See also LEAF Trac ticket #5.

Important: The asterisk executable seems to segfault early in the startup process unless the "-p" command-line argument is specified (or "highpriority = yes" is specified in /etc/asterisk/asterisk.conf).

Packages

The Asterisk distribution is divided into four .lrp Packages, based on the structure adopted for Asterisk 1.2 on Bering-uClibc 3.x but somewhat extended. The Packages are:

asterisk.lrp 
The main Package, with the Asterisk executables installed into directory /usr/sbin/ and also all of the Asterisk modules installed into directory /usr/lib/asterisk/modules/
Note: There are no configuration files included in asterisk.lrp - see the Installation section for more details
astsmpls.lrp 
The sample configuration files which are installed into directory /etc/asterisk/
astsnds.lrp 
The sound files which are installed into directory /var/lib/asterisk/sounds/
This file is big - 1.7MB - and once installed the files occupy over 2.25MB of disk space
astmoh.lrp 
The music-on-hold files which are installed into directory /var/lib/asterisk/moh/
This file is very big - 14.6MB - and once installed the files occupy over 17MB of disk space

The following Packages are pre-requisites for asterisk.lrp:

  • libm.lrp
  • libssl.lrp
  • libcrpto.lrp
  • lpthread.lrp
  • ncurses.lrp

Installation

There are no configuration files included in the asterisk.lrp Package. Sample configuration files are available in the astsmpls.lrp Package and - once present - these are saved (and restored) as part of the basic asterisk.lrp Package. The recommended procedure is therefore to load the astsmpls.lrp Package manually. This only needs to be done once. This can be achieved with the following command (assuming the .lrp files are available in the /mnt/ directory):

apkg -i /mnt/astsmpls.lrp

Note that the sample configuration files are loaded into directory /etc/asterisk/samples/. Those files which you require for your configuration will need to be moved or copied into /etc/asterisk/ before they will be processed by Asterisk.

A Simple Test Configuration

Asterisk is a large and complex application and this page is not intended to be an Asterisk tutorial or reference manual. However, a very simple but working configuration can be achieved with the following steps:

  1. Install the asterisk.lrp and astsmpls.lrp Packages along with any missing pre-requisites.
  2. Copy /etc/asterisk/samples/*.conf to /etc/asterisk/
  3. Edit /etc/asterisk/asterisk.conf and:
    • Remove the comment character at the start of the line for highpriority = yes
  4. Edit /etc/asterisk/sip.conf and:
    • Remove the comment character at the start of the line for match_auth_username=yes
    • Remove the comment character at the start of the line for allowguest=no
    • Add an entry for a "peer" (SIP phone) - something like the following:
      [host-mac_address_without_colons]
      type=friend
      secret=<your password here>
      deny=0.0.0.0/0
      permit=192.168.1.0/24
      qualify=100
      nat=no
      host=dynamic
      directmedia=yes
  5. Ensure that any firewall running on the VOIP server is configured to permit UDP connections on port 5060
  6. Start the asterisk process in verbose mode in the foreground by running the following command:
    asterisk -vvvc

Incomplete but to be continued... Davidmbrooke 20:09, 22 October 2011 (UTC)

Further Reading

The standard Asterisk version 1.8 documentation is located here.



Prev Up Next