Bering-uClibc 5.x - User Guide - Advanced Topics - Setting Up a VOIP Server

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Advanced Topics - Setting Up a VOIP Server
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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 5.x is based on Yate version 3. Since many of these Yate modules will only be of interest to advanced users, the Bering-uClibc 5.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 and only lists the configuration files for these default Yate modules in the lrcfg menu.


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

  • yate.lrp
  • libcxx.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


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                                        

For example, after dialling 99991003@ (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        

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


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.


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.

 description=Test SIP account

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.



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).


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:

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
The sample configuration files which are installed into directory /etc/asterisk/samples/
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
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:

  • libssl.lrp
  • libcrpto.lrp
  • ncurses.lrp


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, astsmpls.lrp and astsnds.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 SIP phone at the end of the file - something like the following:
  5. Configure the SIP phone device to Register with the Asterisk server, using the host-mac_address_without_colons string as Username and <your-password-here> as the Password
  6. Ensure that any firewall running on the VOIP server is configured to permit incoming UDP connections on port 5060
  7. Start the asterisk process in verbose mode in the foreground by running the following command:
    asterisk -vvvc
  8. Restart the SIP phone device and check that it registers with Asterisk
    • Should be a message in the window where Asterisk was started

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

Hints and Tips

Asterisk CLI

The Asterisk server software has a Command-Line Interface (CLI) which is very useful for interactive use (e.g. checking the status of the system) or for issuing commands from an external shell script. To run the CLI in interactive mode:

asterisk -r

To run a single command in non-interactive mode:

asterisk -rx "command"

This is how /etc/init.d/asterisk shuts down Asterisk, for example (it executes asterisk -rx "core stop now").

CLI Commands

Some of the CLI command syntax can be a bit cryptic and it must have changed fairly recently since a lot of the online resources are incorrect. The following seem to work OK for version 1.8.

module show 
lists the Asterisk modules which are currently loaded
sip show peers 
shows the registration state of the SIP "peers"
sip show registry 
shows the state of upstream SIP providers which Asterisk is registered with

Further Reading

  • The standard Asterisk version 1.8 documentation is located here.
  • Asterisk: The Definitive Guide, by Leif Madsen, Jim Van Meggelen, Russell Bryant. 3rd Edition. Copyright 2011 O'Reilly Media, Inc. ISBN 978-0-596-51734-2.

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