ATCOM APBX IP04 User's Guide

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Unit 16:
Unit 16:
VoIP- 4D Primer - Building Voice
VoIP- 4D Primer - Building Voice
Infrastructure in Developing Regions
Infrastructure in Developing Regions
Authors: Alberto Escudero-Pascual - Louise Berthilson
Acknowledgements:
Adel El Zaim (Arabic and French editor), Anas Tawileh (Arabic translator), Iñaki
Cívico y Sylvia Cadena (Spanish editors),
Johan
Bilien (French translator) and Martin Benjamin
(English editor).
Table of Contents
1. About this document ...........................................................................................................................4
1.1 Degrees of difficulty.......................................................................................................................4
1.2 Information about the icons...........................................................................................................4
2. Introduction..........................................................................................................................................4
3. The magic potion.................................................................................................................................6
3.1 VoIP...............................................................................................................................................6
3.2 Open Standards and Free and Open Source Software..................................................................7
3.3 Asterisk..........................................................................................................................................8
4. The recipe............................................................................................................................................9
4.1 PBX................................................................................................................................................9
4.2 PSTN...........................................................................................................................................10
4.3 Signalling in traditional telephony.................................................................................................10
4.3.1 Analogue signalling..................................................................................................................11
4.3.2 Telephone Exchange Signalling..............................................................................................12
4.4 Signalling in IP Telephony ...........................................................................................................12
4.4.1 Session Initiation Protocol (SIP)..............................................................................................12
4.4.2 Proxy Servers..........................................................................................................................13
4.4.3 Real Time Protocol and NAT...................................................................................................13
4.4.4 Inter-Asterisk eXchange (IAX).................................................................................................14
4.5 VoIP Hardware.............................................................................................................................15
4.5.1 VoIP Phones............................................................................................................................15
4.5.2 Soft Phones.............................................................................................................................15
4.5.3 PSTN interface cards..............................................................................................................15
4.5.4 Analogue Telephone Adaptors................................................................................................16
4.6 Codecs.........................................................................................................................................17
4.7 Quality of Service.........................................................................................................................17
Page 1 TRICALCAR | www.wilac.net/tricalcar - Version: February 2008
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Summary of Contents

Page 1 - Table of Contents

Unit 16:Unit 16: VoIP- 4D Primer - Building VoiceVoIP- 4D Primer - Building Voice Infrastructure in Developing RegionsInfrastructure in Develo

Page 2

4.2 PSTNPSTN stands for public switched telephone network, “the telephone network of the telephone networks” or most commonly known as

Page 3

A Foreign Exchange Station (FXS) is what is on the other side of a traditional telephone line. A FXS delivers the dial tone and ring tone to the phon

Page 4 - 2. Introduction

Unfortunately, there are many ways to generate these kind of signals. Each mechanism is known as a “signalling method”. Signalling methods vary from o

Page 5

Three important things that SIP does are :1. Dealing with authentication2. Negotiating the quality15 of the phone call 3. Handling the IP addresses an

Page 6 - 3. The magic potion

Network Address Translators (NATs) are the big enemies of RTP. A NAT-network consists of several computers that share one public IP address with the

Page 7 - USD/month

The concept of trunking can be explained with the following metaphor: Imagine that you need to send five letters to people living

Page 8 - 3.3 Asterisk

4.5.3 PSTN interface cardsIf you want to route calls from your VoIP terminals to the traditional telephone network (PSTN) you need to include dedicate

Page 9 - 4. The recipe

If you want to implement a network in a developing region you might consider using ATAs as they are normally cheaper than VoIP phones. ATAs are also

Page 10 - 4.2 PSTN

4.7 Quality of ServiceQuality of Service (QoS) is the ability of a network to provide improved (better) service to selected

Page 11 - 4.3.1 Analogue signalling

VoIP appliances will allow you to set a “jitter buffer.” A jitter buffer is a shared data area where voice packets can be collected, stored, and sent

Page 12

4.7.1 Latency...184.7

Page 13 - 4.4.2 Proxy Servers

5.1.1 Installation TipsThis is a list of installation tips:If you need to install a TDM400P card• Be sure that your PC has an empty 2.2 PCI slot.• If

Page 14 - Asterisk

The approach we are taking in this short guide is not to include a listing of all possible commands but rather to walk through three basic scenarios w

Page 15 - 4.5 VoIP Hardware

6.2 Fetching Asterisk PackagesIt is also possible to get Asterisk pre-compiled. Pre-compiled software comes in the form of a “p

Page 16 - 4.5.3 PSTN interface cards

also do far more things that you do not expect. By keeping things simple you will learn slowly but steadily. 6.4 Basic Asterisk CommandsAsterisk ha

Page 17 - 4.6 Codecs

6.5 Configuration FilesThe number of configuration files that you need to edit in order to run Asterisk depends on the types of VoIP technologies that

Page 18 - 4.7 Quality of Service

Configuration file Description(always mandatory)/etc/asterisk/sip.conf Used to configure SIP based channels (SIP VoIP phones and SIP providers)/etc/

Page 19 - 5.1 What do I need?

VIA or AMD Geode processors, these platforms have their limitations in terms of computational power and cost.In May 2007, I discovered the Free Teleph

Page 20 - 6. Installing Asterisk

During the time of writing (September 2007), the unit can be configured editing the configuration files via a serial cable or the network interface. T

Page 21 - 6.1 Compiling Asterisk

Parameter SettingIP address of the VoIP phone 192.168.46.2IP address of the SIP Proxy (our PBX) 192.168.46.1Registration YESUser name/Auth name 462Cal

Page 22

Instead of using a traditional fixed phone, we decide to connect a cordless DECT phone base station35 to the RJ-11 port of the ATA box. The result is

Page 23 - 6.4 Basic Asterisk Commands

12.2 Limiting factors for IP telephony disemmination...4612.3 The jargon...

Page 24 - 6.5 Configuration Files

[iaxy_school]registerAssume that your local DHCP server assigns the IP address 192.168.46.100 to the IAXy. Then, from the Asterisk console run the fo

Page 25 - 6.6 Peers, Users and Friends

• Enable the possibility of receiving audio by the same port that we send audio. In the PBX:• Inform Asterisk that the softphone is inside of a NAT.A

Page 26

In sip.conf, the following data should be added: [462]type=friend ; We can call and receive callssecret=462passcontext=internal_calls ;

Page 27 - 7.1 Background

qualify=yes ; We send dummy traffic to keep the NAT openIn iax.conf , the following data should be added:[464]type=friendsecret=464passcontext=inter

Page 28 - 7.2.2 Hospital

exten => 464,1,Dial(IAX2/464)exten => 466,1,Dial(IAX2/466)exten => t,1,Hangup() ; Special extension (Timeout)exten => i,1,Hangup() ; Spe

Page 29 - 7.2.3 Primary School

The second step is to ensure that the hardware drivers have been properly compiled and loaded. By running #lsmod you should see that the wctdm dri

Page 30 - • Enable keep-alive packets

Channel 01: FXS Loopstart (Default) (Slaves: 01)1 channels configured.Step 4: Configuring Asterisk to use the Zapata HardwareThe forth and final

Page 31 - 7.3 Configuring Asterisk

exten => s,1,Answer() ; We answer the callexten => s,2,DigitTimeout(10) ; Setting Timeout values in secondsexten => s,3,ResponseTime

Page 32

The process is simple; after powering off the PBX, we plug an FXS module into the second port of the TDM card. After powering on the syst

Page 33

echotraining=yescontext=incoming_pstnsignalling=fxs_lschannel => 1context=internal_callssignalling=fxo_lschannel => 2Image 10. The Telecentre us

Page 34

1. About this document This material is part of the course package created for TRICALCAR project. For information on TRICAL

Page 35

exten => t,1,Hangup()include => internal_calls[internal_calls]exten => 461,1,Dial(Zap/2) ; Extension 461 calls via Zap channel 2exten =&

Page 36 - [incoming_pstn] as follows:

Image 11: The Telecentre and the Training Centre are equipped with one PBX each. The PBX's are Interconnected via a VSAT satellite link.9.1 Typi

Page 37 - [internal_calls] section:

tos = lowdelaydisallow = allallow = ulawallow = g729 ; We add the G.729 codecregister => server2:server2pass@training_voip.org; server2:server2pas

Page 38

exten => s,1,Dial(Zap/2) ; Calls from the Training Centre ring ; the Telecentre Phone9.2.2 Training CentreThe iax.conf configuration file in the

Page 39 - 8.3 Updating the Dialplan

In the Training Centre we decide that all calls coming from any Telecentre are forwarded using SIP to the support desk.[incoming_telecentres_calls]ext

Page 40

services to digitally excluded areas, while promoting the creation of community operated and managed telephone networks.The samples files that we incl

Page 41 - 9.2.1 Telecentre

During the review we have looked into three major areas:• Specialized Asterisk software distributions including graphical configuration tools (The Ast

Page 42

• Many of the projects and tools have changed name during the last years and it is difficult to understand what exactly each of the tools is providing

Page 43 - 9.2.2 Training Centre

12.4.1 FreePBX It is a PBX framework build at the top of asterisk that includes a GUI to manage a asterisk based telephone system. FreePBX grew up in

Page 44 - 11. Conclusions

12.6 Other GUIs12.6.1 SwitchboxA company recently acquired by Digium (the company behind Asterisk) that has released the Swi

Page 45 - 12.1 Introduction

Wireless technologies come in many flavors and their performance depends on the particular application.It was not until spri

Page 46

The following table summarizes our findings and positions the IP04 from the Free Telephony Project as the most mature low cost solution for embedded t

Page 47 - The jargon

A big step towards the dissemination of IP Telephony can be found in the Free Telephony Project, a community project that has developed an open hardwa

Page 48 - 12.5 AsteriskGUI

13.2 The IP04 Open Hardware IP-PBXThe IP04 is a 4 port IP-PBX that runs Asterisk and uClinux on a powerful embedded Bl

Page 49 - 12.6 Other GUIs

for Linux. He has a broad range of telephony hardware, software, DSP, and management experience and has held executive level positions in the sat-com

Page 50 - 12.9 Conclusions

13.3.1 Community development modelIt should be stressed that the IP04 is a community effort, with many people contributing. In no

Page 51 - 13.1 Executive Summary

Some changes to Asterisk were required to account for the lack of FPU and MMU on the Blackfin, for example porting of DTMF routines from floating poin

Page 52

Bug counts have been very low and development cycles very fast due to re-use of existing open hardware modules. The IP04 was booting uClinux and

Page 53

same models can apply to the developing world. In fact VOIP models may offer GSM providers opportunities to expand their network and bill

Page 54 - 13.4 IP04 Design

• Communication within organisations. Consider a university which has WiFi based Intranet but no fixed landlines. Rather than using GSM handset for in

Page 55 - 13.4.1 Open Hardware Design

14. Table of AbbreviationsAbbreviation DescriptionATA Analogue Telephone AdapterDECT Digital Enhanced Cordless TelecommunicationsFXO Foreign Exchange

Page 56 - 13.5 Competitor Analysis

• Private telephony in a rural community (Section 6)• Connecting the local telephone network to the PSTN (Section 7)• Interconnecting two remote commu

Page 57

1st Edition (December 2006)16. Description of illustrationsPBX running AsteriskATAIAXYWireless router Wireless bridgeSwitchRouterAnalogue Phone DECT P

Page 58 - 13.8 Next Steps

SoftphoneParabolic antennaVSATSatellite17. Intellectual Property RightsThe materials developed for the TRICALCAR project utilise a short version of th

Page 59 - 15. Document changes

• Share Alike. If you alter, transform, or build upon this work, you may distribute the resulting work only under the same or similar license to this

Page 60

bandwidth is highly limited and the costs for Internet access are at least 100 times higher than in Europe and North America.8To get a sense of how b

Page 61

To understand why open standards are important, let us start by providing a simple definition of a “standard.” A standard is a set of rules, condition

Page 62

4. The recipeThis section summarizes the basic concepts behind VoIP. Understanding each of them will be very useful when you start configuring an

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