trixbox2_without_tea.. - UV UTBM J. Millet - Free

TRIXBOX-2 without Tears 

Incorporating PBX in a Flash 

The ICT serial following 

Trixbox without Tears 

If you find this book helpful and you are a private user, a small PayPal donation made to 

bensharif@gmail.com would be very highly appreciated. If you are in Australia and don’t have Paypal, you may 

forward a cheque made to: 

Ben Sharif, PO Box 70 Kingswood NSW 2747, Australia. 

(For business use or using this as manual for your customers, prior arrangement with the author is required) 

http://dumbme.voipeye.com.au/Trixbox/index1.htm 

TRIXBOX-2 Without Tears Page 1 of 238

Intentionally left blank 


For TRIXBOX 2.x with freePBX 

(Version 2.6.0a) 

(Work in Progress is Perpetually ongoing) 

by 

Ben Sharif 

(The same author who brought you Asterisk@Home without tears and Trixbox without tears) 

About the Author 

Last updated on April 21, 2008 

Please send comment or suggestions to me at the address below 

 

mailto:netvoice.solutions@gmail.com 

First Published on 1 st January 2007 

This document is released under the Creative Commons By-Attrib Non-Commercial Share-Alike 2.1 licence. 

This work is dedicated to my wife, who has been very patient with me, Andrew Gillis, Ward 

Mundy and Rob Thomas whose devotions to TRIXBOX, PiaF and freePBX made it all 

possible. 

* 

Stay tuned for our next episode – TRIXBOX version X 


Intentionally left blank 


TABLE OF CONTENT 

Forward......................................................................................................................................... 11 

1 Introduction.............................................................................................................................. 12 

1.1 Is VoIP for you............................................................................................................ 12 

1.1.1 What is it going to cost........................................................................................ 12 

1.1.2 What will the quality of the phone calls be.......................................................... 12 

1.2 What is Trixbox........................................................................................................... 12 

1.3 The IP PBX Components ............................................................................................. 13 

1.3.1 The IP PBX ........................................................................................................... 13 

1.3.2 Phones.................................................................................................................. 14 

1.3.3 SIP Gateway ......................................................................................................... 14 

1.3.4 Home Network ...................................................................................................... 14 

1.3.5 VOIP Service Providers ........................................................................................ 14 

2 Initial Preparation and Planning............................................................................................. 15 

3 Installation................................................................................................................................ 16 

3.1 Initial Installation........................................................................................................... 16 

3.2 Change Default Settings .............................................................................................. 19 

3.2.1 To get Help ........................................................................................................... 20 

3.2.2 Change IP Address (set IP address to Static) ...................................................... 20 

4 Set-up TRIXBOX....................................................................................................................... 23 

4.1 Installing Trixbox Modules............................................................................................ 24 

4.2 Installing freePBX Modules .......................................................................................... 26 

5 Basic configuration using freePBX ....................................................................................... 30 

5.1 General Settings........................................................................................................... 30 

5.1.1 Dial Command Options......................................................................................... 31 

5.2 Extensions.................................................................................................................... 34 

5.2.1 Create Extensions................................................................................................. 34 

5.2.2 Enable Voicemail ................................................................................................... 36 

5.3 Follow Me ..................................................................................................................... 37 

5.4 Ring Groups ................................................................................................................. 38 

6 Set up softphones. .................................................................................................................. 40 

6.1 CounterPath X-Lite Softphone ..................................................................................... 40 

6.2 BOL SIPPhone............................................................................................................. 41 

Profile Tab....................................................................................................................... 41 

Audio & Video Tab .......................................................................................................... 42 

Network Tab.................................................................................................................... 42 

STUN Tab ....................................................................................................................... 42 

Call Forward.................................................................................................................... 43 

7 Trunks to the outside world ................................................................................................... 44 

7.1 What is a Dial Pattern ................................................................................................ 44 

7.2 What is a Trunk .......................................................................................................... 44 

7.3 Trunks and Outbound Routes – a circus of confusion................................................. 45 

7.3.1 Outgoing Route Dial Pattern................................................................................. 45 

7.3.2 Trunk Outgoing Dial Rules.................................................................................... 45 

7.4 Let’s Create The VoIP Trunks...................................................................................... 46 

7.4.1 Pennytel ................................................................................................................ 46 

7.4.2 Oztell (SIP)............................................................................................................ 48 

6.4.3 Oztell (IAX)............................................................................................................ 49 

8 Calls Handling.......................................................................................................................... 51 

8.1 Inbound Routes (Incoming Calls)................................................................................. 51 

8.1.1 Inbound Route from ZAP Trunk............................................................................ 52 

8.2 Time Conditionds ......................................................................................................... 53 

8.2.1 Simple Time Condition.......................................................................................... 54 

8.2.2 Complex Time Condition....................................................................................... 54 

8.3 Outbound Routes ......................................................................................................... 55 

8.3.1 What is an Outbound Route ............................................................................... 55 

8.3.2 How does it work................................................................................................. 55 

8.3.3 International .......................................................................................................... 56 

8.3.4 Domestic ............................................................................................................... 57 


8.3.5 MobileAust ............................................................................................................ 58 

8.3.6 Oztellonly .............................................................................................................. 58 

8.4 ENUM........................................................................................................................... 59 

8.4.1 Setting up ENUM Trunk........................................................................................ 59 

8.4.2 Setting up ENUM Outbound Route....................................................................... 60 

9 Custom Contexts Module ...................................................................................................... 61 

Downloading and unpacking module. ................................................................................. 61 

Installing Custom Context Module ...................................................................................... 61 

Add and Configure New Custom Context ........................................................................... 62 

Apply a Custom Context to Extension................................................................................. 63 

10 System Recordings ............................................................................................................... 64 

10.1 Direct recording.......................................................................................................... 65 

10.2 Upload Pre-recorded Material .................................................................................... 65 

11 Digital Receptionist ............................................................................................................... 67 

11.1 Setting Up Digital Receptionist................................................................................... 67 

11.2 Multi-Language IVR.................................................................................................... 70 

11.2.1 Complimentary multi-language MOH.................................................................. 71 

11.3 Multi-Tenants IVR....................................................................................................... 72 

12 Get Under The Bonnet .......................................................................................................... 74 

12.1 Editing The .conf Files................................................................................................ 75 

12.1.1 sip.conf................................................................................................................ 75 

12.1.2 sip_custom.conf.................................................................................................. 76 

12.1.3 iax.conf................................................................................................................ 76 

12.1.4 iax_custom.conf.................................................................................................. 76 

12.1.5 Indications.conf................................................................................................... 76 

12.1.6 enum.conf ........................................................................................................... 76 

12.1.7 extensions_custom.conf ..................................................................................... 77 

12.1.8 features.conf ....................................................................................................... 77 

13 Check your System ............................................................................................................... 79 

13.1 System Process Status .............................................................................................. 79 

13.2 Asterisk Info................................................................................................................ 80 

14 Interfacing Asterisk to PSTN ................................................................................................ 82 

14.1 A100P FXO PCI Card ................................................................................................ 82 

14.2 Create a ZAP trunk..................................................................................................... 83 

14.2.1 zapata.conf ......................................................................................................... 83 

14.2.2 zapata-auto.conf ................................................................................................. 83 

14.2.3 zaptel.conf........................................................................................................... 83 

14.3 OpenVox A400P FXO/FXS Card ............................................................................... 84 


14.3.2 modprobe.conf.................................................................................................... 85 

14.4 Is the A100P/A400P installed................................................................................... 86 

14.5 Caller ID (CID) using ZAP device............................................................................... 87 

14.5.1 zapata.conf ......................................................................................................... 87 


14.6 Sipura SPA3000/3102 as a PSTN Interface .............................................................. 88 

14.6.1 Log in to SPA3000/3102..................................................................................... 88 

14.6.2 Change the settings............................................................................................ 88 

14.6.3 Add SIP Trunk..................................................................................................... 90 

14.6.4 SPA3000 as an outbound PSTN Trunk.............................................................. 91 

14.7 SPA3000 remote management................................................................................... 91 

14.8 Use STUN Server on SPA3000 ................................................................................. 92 

15 PSTN to VOIP Gateway ......................................................................................................... 93 

16 DISA ........................................................................................................................................ 94 

16.1 DISA - Hand Coding Method...................................................................................... 94 

16.2 DISA – Using freePBX ............................................................................................... 95 

17 Operator Panel....................................................................................................................... 96 

17.1 Flash Operator Panel (FOP) ...................................................................................... 96 

17.1.1 Setting the Admin Password............................................................................... 97 

17.1.2 Hang-up a Call .................................................................................................... 97 

17.1.3 Transfer a Call .................................................................................................... 97 


17.1.4 Initiate a Call ....................................................................................................... 97 

17.1.5 Barge in or Create a Conference........................................................................ 97 

17.2 HUDLite...................................................................................................................... 98 

17.2.1 Setting up HUDLite Admin.................................................................................. 98 

17.2.2 Setting Up HUDLite............................................................................................. 99 

17.2.3 What if it does not work .................................................................................. 102 

18 Call parking and transfer .................................................................................................... 103 

18.1 Call Transfer - Managed .......................................................................................... 103 

parking_additional.inc ................................................................................................... 103 

18.1.1 How is it done ................................................................................................. 104 

18.2 Call Transfer – Blind................................................................................................. 105 

18.3 Put a Call On Hold.................................................................................................... 105 

18.4 Call Pickup ............................................................................................................... 105 

18.5 Filter Your Incoming Calls - Only Accept Known Calls ............................................ 105 

19 Conference ........................................................................................................................... 108 

meetme.conf ................................................................................................................. 108 

meetme_additional.conf................................................................................................ 108 

19.1 Create Conference.................................................................................................... 108 

19.2 Invalid Conference number ...................................................................................... 109 

20 Voicemail .............................................................................................................................. 111 

20.1 Enabling Email Notification ...................................................................................... 111 

20.1.1 Installing sendmail ............................................................................................ 111 

20.1.2 /etc/hosts........................................................................................................... 112 

20.2 Use Postfix Instead of Sendmail .............................................................................. 114 

20.3 How to Retrieve Voicemail from External Phone ..................................................... 114 

20.3.1 Method 1 ........................................................................................................... 114 

20.3.2 Method 2 ........................................................................................................... 115 

20.4 How to Change Voice Mail Greetings ...................................................................... 115 

21 Fax to E-Mail......................................................................................................................... 116 

21.1 Installing Fax for TRIXBOX ...................................................................................... 116 

22 Weather Reports.................................................................................................................. 117 

22.1 Configure Weather Report – On Demand................................................................ 117 

22.1.1 Create Extension codes.................................................................................... 118 

22.2 Configure Weather Report – Background Method................................................... 118 

22.2.1 Create Extension codes.................................................................................... 119 

22.3 Now put it in your Digital Receptionist...................................................................... 120 

23 Remote Management .......................................................................................................... 121 

23.1 httpd.conf.................................................................................................................. 121 

24 How to set up a Remote Extension.................................................................................... 122 

24.1 Create a new extension ........................................................................................... 122 

24.2 IAX.Conf ................................................................................................................... 123 

24.3 Sip_Nat.Conf............................................................................................................ 123 

24.4 Setting up an IAX Softphone.................................................................................... 123 

24.4.1 First the general options ................................................................................... 124 

24.4.2 Accounts options............................................................................................... 125 

25 Calls Recordings ................................................................................................................. 126 

25.1 ARI Interface ............................................................................................................ 126 

25.1 Record all calls by default ........................................................................................ 127 

25.2 Record calls on demand. ......................................................................................... 127 

26 STUN Servers....................................................................................................................... 128 

27 Dialing through MS Outlook® ............................................................................................ 128 

27.1 Download AstTapi .................................................................................................... 128 

27.2 Install AstTapi........................................................................................................... 129 

27.3 Configure Outlook Address Book............................................................................. 129 

28 Speed Dial ............................................................................................................................ 131 

28.1 The Dumb-Me method ............................................................................................. 131 

28.2 TRIXBOX’s method (Version 1.x) ............................................................................ 132 

28.3 TRIXBOX’s method (Version 2.x) ............................................................................ 132 

29 How to use Window Messenger® 5.x ................................................................................ 134 

30 How to interconnect 2 boxes.............................................................................................. 136 


30.1 Method 1 - With the Peer Asterisk box as Extensions............................................. 136 

30.2 Method 2 - In a Peer/User Arrangement.................................................................. 137 

30.3 Method 3 – Friend/Friend Arrangement................................................................... 138 

30.4 Thinking of more than 2 boxes............................................................................... 139 

30.5 Redirect Incoming in Box A to Box B ....................................................................... 139 

31 Customised Voice ............................................................................................................... 141 

31.1 Customising Voice Prompts ..................................................................................... 141 

31.2 Installing Other Languages ...................................................................................... 142 

31.3 Converting WAV files ............................................................................................... 142 

31.3.1 Converting WAV to GSM .................................................................................. 142 

31.3.2 Converting WAV to SLN ................................................................................... 143 

32 MOH (Music On Hold).......................................................................................................... 144 

32.1 Customising MOH .................................................................................................... 144 

32.2 MOH for different trunks........................................................................................... 145 

32.3 Streaming Music on Hold ......................................................................................... 146 

Method 1 ....................................................................................................................... 146 

Method 2 ....................................................................................................................... 147 

33 Backup and restore ............................................................................................................. 149 

33.1 Schedule a Backup .................................................................................................. 149 

33.2 To Restore a Backup................................................................................................ 151 

34 SugarCRM ............................................................................................................................ 152 

35 USB Phone Support ............................................................................................................ 154 

36 Codec (Coder Decoder) ...................................................................................................... 155 

36.1 Installing G729 and G723 CODECs......................................................................... 156 

36.2 To Install CODECS on PiaF...................................................................................... 157 

37 PSTN Trunk – Special Handling......................................................................................... 159 

37.1 Dial 0 for Outside Line.............................................................................................. 159 

37.1.1 Using Zaptel Card Method................................................................................ 159 

37.1.2 Using SPA3K Method ....................................................................................... 159 

37.2 Giving Priority to Emergency Numbers (E.g. 000) ................................................... 160 

38 Port Forwarding and QOS – Routers................................................................................. 161 

38.1 Port Forwarding........................................................................................................ 161 

38.2 QOS – Routers......................................................................................................... 161 

39 Upgrading your system ...................................................................................................... 163 

39.1 Upgrading Trixbox.................................................................................................... 163 

39.2 Upgrading freePBX .................................................................................................. 163 

40 Tools ..................................................................................................................................... 164 

40.1 Webmin for Trixbox .................................................................................................. 164 

40.1.1 To update WebMin............................................................................................ 165 

40.3 Putty ......................................................................................................................... 165 

40.4 WinSCP.................................................................................................................... 165 

41 Use ACCESS to connect to Database ............................................................................... 167 

41.1 Setting up a new MySQL user ................................................................................. 167 

41.2 Configure MySQL ODBC for Windows .................................................................... 168 

41.3 Connecting using MSACCESS ................................................................................ 170 

42 My TRIXBOX® PBX Network ............................................................................................. 174 

43 Trouble Shooting................................................................................................................. 175 

43.1 Debug Messages and Log Files are your friends .................................................... 175 

43.2 Unable to log on to Packages ................................................................................ 176 

43.3 Asterisk Feature Codes not working ........................................................................ 176 

43.4 Asterisk Drops Calls after a few seconds ................................................................ 176 

43.5 Pennytel Asterisk Problem ....................................................................................... 176 

43.6 Unable to receive Incoming Calls............................................................................. 176 

43.7 Sipura SPA3000 – Not hanging up after PSTN Calls .............................................. 177 

43.8 System Not Working after changing to Static IP ..................................................... 177 

43.9 No Audio on Analog line........................................................................................... 178 

43.10 Unexplained Crashes............................................................................................. 178 

43.11 ZAP Inbound Route Does not Work....................................................................... 178 

43.12 Occassional FATAL Error when defining extension............................................... 178 

43.13 All extensions failed to work when Internet down .................................................. 179 


Appendix A................................................................................................................................. 180 

A.1 Asterisk Feature Codes ............................................................................................. 180 

A.2 Asterisk CLI commands............................................................................................. 181 

A.2.1 General commands ............................................................................................ 181 

A.2.2 AGI Commands .................................................................................................. 182 

A.2.3 Database Handling............................................................................................. 182 

A.2.4 IAX Channel Commands.................................................................................... 182 

A.2.5 SIP Channel commands..................................................................................... 183 

A.2.6 Server management........................................................................................... 183 

A.3 Asterisk Special Extensions....................................................................................... 183 

A.4 Asterisk Common Variables ...................................................................................... 184 

A.5 Indications for Various Countries............................................................................... 185 

Appendix B................................................................................................................................. 194 

B.1 – Australian VSPs (trunks).......................................................................................... 194 

B.1.1 BBPglobal........................................................................................................... 194 

B.1.2 Engin BYO.......................................................................................................... 194 

B.1.3 Faktortel (IAX) .................................................................................................... 195 

B.1.4 iPrimus................................................................................................................ 195 

B.1.5 Koala................................................................................................................... 196 

B.1.6 MyNetFone ......................................................................................................... 196 

B.1.7 Nehos (IAX) ........................................................................................................ 197 

B.1.8 Nodephone ......................................................................................................... 197 

B.1.9 SipMe.................................................................................................................. 197 

B.2 – New Zealand VSPs (trunks) .................................................................................... 198 

B.2.1 iTalk .................................................................................................................... 198 

B.2.2 Double Zero 64 (IAX)......................................................................................... 198 

B.3 – Other VSPs (trunks)................................................................................................. 199 

B.3.1 Broadvoice.......................................................................................................... 199 

B.3.2 FWD aka Pulver(IAX) ......................................................................................... 199 

B.3.3 SipBroker............................................................................................................ 199 

B.3.4 SipPhone ............................................................................................................ 200 

B.3.5 VOIP Buster........................................................................................................ 200 

Appendix C................................................................................................................................. 201 

C.1 Voice for Asterisk Messages ..................................................................................... 201 

C.2 Australian Voice and Language Set for Asterisk ....................................................... 202 

C.3 More Australian Voice for Asterisk ............................................................................ 202 

Appendix D (ATAs and Routers).............................................................................................. 203 

D.1 Alcatel SpeedTouch 510/530 ADSL Router .............................................................. 203 

D.2 Billion 7402 VL........................................................................................................... 203 

D.3 Linksys PAP2 and Sipura SPA2000.......................................................................... 204 

Appendix E (Users’ Contributions).......................................................................................... 206 

E.1 Suggestions for Dial Plan and configuration.............................................................. 206 

E.3 Eliminating Echo Problems in SPA-3000 .................................................................. 209 

E.4 Immediate Answer of PSTN Calls ............................................................................. 211 

E.5 Rebuild Zaptel Driver (Users’ Suggestions) ............................................................. 211 

E.5.1 As experienced by marner – a Whirlpool Forum participant. ............................. 211 

E.5.2 As suggested by Rob Thomas (the FreePBX guru)........................................... 211 

E.6 Zaptel card configuration for Australia....................................................................... 212 

E.7 RAID1, LVM (logical volumes) – how to .................................................................... 214 

E.8 V2.2 Trixbox Load Config to include IDE raid............................................................ 223 

E.9 TEST Bench Anyone ............................................................................................... 226 

Appendix F (Publications and References) ........................................................................... 227 

F.1 DUNDi – a guide to a Simple Setup........................................................................... 227 

F.2 The future of Telephony by O’Reilly Publishing......................................................... 227 

F.3 Other Asterisk Forums, Tutorials and Wikis ............................................................. 227 

F.4 Voice over IP – Per call bandwidth consumption....................................................... 227 

Notes........................................................................................................................................... 228 

A2Billing............................................................................................................................. 228 

Delete Call Records from CDR ......................................................................................... 228 

Extension Numbers to avoid using.................................................................................... 229 


sip_nat.conf ....................................................................................................................... 229 

How to restrict Outgoing Calls – Manual Method.............................................................. 229 

Bugs Reports ............................................................................................................................. 233 

Asterisk re-load issue........................................................................................................ 233 

Are you still having problem ............................................................................................ 233 

Hangs on shutdown........................................................................................................... 233 

Zaptel Not Detected on Trixbox 2.2.4 ............................................................................... 234 

All phones UNUSABLE if internet connection is lost. ....................................................... 234 

Cheat Sheet ................................................................................................................................ 236 

Some Linux commands in common use ........................................................................... 236 

Bibliography............................................................................................................................... 237 

Changes Made since last Update............................................................................................. 238 


FORWARD 

Following the tradition of Asterisk@Home without Tears and Trixbox without Tears, this 

guide is not meant to be a detailed manual for Trixbox®; instead it is just a quick and dirty 

guide and a resource to assist you get started with minimum pain. As always, use this 

document only as a guide, not as a Gospel. It does not cover every detail in Trixbox or 

PiaF neither will it provide you with a complete solution or spiritual guidance. 

Only a competent VoIP consultant can provide you the necessary know-how for a 

complete solution, and as for spiritual guidance, I will leave that entirely to you. 

This document was compiled mainly based on Australian conditions, through my 

experience going through the wash, supplemented by inputs from dozens of participants 

in the Whirlpool and other Trixbox and Asterisk related forums. These people were 

mentioned in A@H and Trixbox without tears. They have helped me whenever I get into 

difficulties, and I still seek guidance (the Trixbox/PiaF/Asterisk type) from them. 

I take no primary credit for this document (I am only the compiler of the not so divine 

revelation). I invite others to participate in improving this document. Please send me your 

feedbacks; so together we can make this document a valuable resource for our peers in 

the Trixbox/PiaF/freePBX community, experienced and new alike. 

- Ben Sharif January 1, 2007. 

 

 

The usual disclaimer: (Don’t you just hate it, having to say this.) 

In its Basic form, all that I have outlined worked for me but it may contain many 

errors – typographical, omissions and even misguidance that may require a lot of 

extra corrections. While I have taken due care, however, it may not necessarily be 

suitable for your individual configuration. I do not take; neither will I accept any 

responsibility for any losses incurred due to actions or inaction conducted as a result 

of methods or advice found in this document. 

I do not provide Trixbox/PiaF/freePBX support and neither do DIGIUM. If 

assistance is required, please post in the forum. Participants in the various 

forums may have a number of solutions and at the same time providing the 

opportunity for others with similar questions to benefit from the solutions 

provided. 

http://forums.whirlpool.net.au/forum-threads.cfmf=107 

Ever since the release of A@H without Tears, I have received numerous emails asking 

for assistance. Since I am putting a lot of efforts into providing personal assistance to 

those who have sent me donations (incidentally, my only source of income), the volume 

of email I received has become unmanageable. While I will try to assist within my limited 

knowledge, I hope you will understand if I don’t answer each and every email. 

 

I am not an expert and therefore I do not have all the answers. I ask those 

requiring assistance to post their questions in the Whirlpool, Digium, freePBX, 

PiaF or Trixbox Forum (forum participants may not answer any question if the 

solution is already in this guide) where there are many knowledgeable Asterisk 

and Linux savvy participants who will be able to assist. 


1 INTRODUCTION 

Asterisk TM is a Linux based IPBX application developed by Mark Spencer of Digium, 

the company behind Asterisk. Trixbox and PiaF TM (PiAF is Ward Mundy’s PBX in a 

Flash) evolved from the core Asterisk. It is made up of several major components. These 

were developed under GPL supported relatively by users themselves. It consists of 

applications, a provisioning system, an installer, and an operating system that, together, 

make a complete package ready for use as an out of the box PBX. 

Within this document, Trixbox, PiaF and Asterisk will be referred to frequently and they 

are interchangeable as Trixbox and PiaF are in essence supersets of the Asterisk. 

1.1 IS VOIP FOR YOU 

Whether VOIP is for you or not depends on a number of or combinations of factors. 

Some economic and quality requirements should be considered. 

1.1.1 What is it going to cost 

Assuming that you already have a broadband service, a router, and a Windows PC to 

run the softphone, the cost will be minimal. 

If you already have a spare computer to dedicate to this task, then the cost is almost 

nothing unless you need to buy an audio headset ($15.00 from your local electronic 

store) for the softphone. If you do not have a spare PC with the above specification, 

then you may be able to buy one from your local swap meets for under $200.00, 

which may include a monitor. Ensure that the PC has an Ethernet NIC for connecting 

to your home network. 

Your only other initial cost will be the $20.00 or so activation fee to Oztell (or other 

VSP of your choice). In fact many VSPs, the like of Pennytel, Primus etc., will 

register you for free if you want to restrict your calls to VoIP only within the same 

VSPs. If you want the ability to make to PSTN or between different VSPs, you may 

have to pay for the calls. 

All these “Major Expenses” will be recovered when you receive your monthly Telstra 

or Optus phone bills. 

1.1.2 What will the quality of the phone calls be 

If you are expecting the quality to be consistently as good as your existing PSTN 

calls, you will be somewhat disappointed, but if you will be happy with a quality that is 

comparable to your mobile phone, not quite but close to your existing PSTN calls and 

also be prepared for the whining by the other half, the phone calls are not being as 

good, while you are leaching off the P2P, you might be in luck. 

VoIP via the Public Internet is very much dependant on a number of factors – 

available bandwidth not withstanding, your usage habit of the internet and LAN traffic 

and equipment quality, amongst others, also play very important roles. 

For further information please refer to the link below: 

http://forums.whirlpool.net.au/index.cfma=wiki&tag=VOIP_Quality 

1.2 WHAT IS TRIXBOX 

Trixbox is a collection of telecommunication utilities and tools compiled together to 

become an integrated IP PBX. The major components that make up Trixbox are; 


• Asterisk, the core PBX 

• Sugar, a CRM system 

• A2Billing – Calling Card platform 

• Flash Operator Panel, a screen-based operator’s console 

• Web Meet Me Control, a meet me conferencing control application 

• freePBX a web-based User Interface tool for Trixbox. 

• A report system – the part of freePBX that provides CDR reporting. 

• A Maintenance system, also part of Trixbox, which provides low level interfaces 

to some components and real time system information 

• CentOS R , a version of Linux related to a very well known Enterprise Linux (but 

without the branding and support). 

1.3 THE IP PBX COMPONENTS 

Four main components needed to set up VoIP PBX: 

• Trixbox, the Asterisk powered IP PBX 

• The phones (or softphones) 

• The VOIP gateway services that let you call other VOIP users and people on the 

PSTN. 

• Have a home network and broadband access with a router and hubs/switches if 

needed. If you are restricting the usage to PSTN only, you will not need the 

broadband, but then it will defeat the whole purpose. 

Trixbox-2 while essentially similar to the previous version 1.x, does consist some set-up 

differences that warrant explanation. I will attempt to describe the set-up of each of 

these. In most cases it does not matter if you are behind a firewall provided that you have 

all the appropriate ports forwarded correctly. You are not required to do anything special 

like running the IP PBX outside the NAT or in the DMZ. However, it is important to 

ensure that you have enough bandwidth (upstream and downstream) to carry voice 

traffic. 

1.3.1 The IP PBX 

You will need a computer to run the IP PBX. It is recommended that you run a 

dedicated PC for this purpose. The PC described below (minimum) will be sufficient 

to power the IP PBX in a small office or home environment. Therefore don’t throw 

away that old Pentium II clunker you have in the attic. 

• 350Mhz Pentium II PC or better (PIII 800 will give extra comfort). 

• 256MB RAM – the more the better 

• 4 GB Hard disk space - minimum. 

• 10/100 NIC 

• CD-ROM Drive 

• 10/100 4 or 8 ports Ethernet hub/switch (not required if your router has spare 

ports) 

 

If you are using Telstra Alcatel SpeedTouch 510/530 ADSL Router, you will need 

to unbind port 5060. To do this, please refer to this link (or see Appendix D). 

http://www.sipgate.co.uk/faq/index.phpaktion=artikel&type=faq&rubrik=024&id=398&lang=de 

When you install Trixbox on this “old” computer, it will take it over – it starts by 

formatting all the hard disks (if you happen to have more than one), so make sure 

there is nothing on the machine that you want to keep. Detailed instructions are 

given in the installation section. 


1.3.2 Phones 

You can buy SIP phones from Grandstream, SNOM, and Cisco etc. Or you can buy 

an ATA (Analogue Telephone Adapter) produced by Linksys, Netcomm or Sipura and 

a few others, which enables you to use an analogue phone (the Linksys PAP2 will 

allow you to hang two analogue phones to it). The Netcomm and Linksys ATAs can 

be bought from Harris Technology and a number of VSPs. 

Register with a VSP such as Faktortel, Oztell, Pennytel and several other VOIP 

Service Providers (VSPs) or you may buy Engin locked ATAs from Harvey Norman 

and Dick Smith Electronic. The choice is yours. 

To get started, it is easiest to get a softphone and run it on another computer. See 

the section on installation of Softphone later in this document. 

1.3.3 SIP Gateway 

It is most likely that you want to communicate with others on the PSTN network, so 

you need to obtain a VOIP gateway service. Since part of the call is being carried on 

the circuit switched network, it costs real money. This means that you will have to 

pay for this part of the system. 

Setting up gateways will be covered in the installation section below. 

1.3.4 Home Network 

The fact that you are reading this and contemplating on installing Trixbox , I assume 

that you have a home network, a broadband service, and are set up behind a Router 

of some sort. 

You will need to pick a static IP address for your IP PBX that is on your home network 

e.g. 192.168.0.100. To keep it simple, please ensure that the Asterisk box is in the 

same network segment of your existing LAN. 

1.3.5 VOIP Service Providers 

If you are in Australia, the Whirlpool VOIP forum has a list of VOIP providers in 

Australia at the following link: http://forums.whirlpool.net.au/forum-replies.cfmt=370536 


2 INITIAL PREPARATION AND PLANNING 

Before setting up your IP PBX, it is recommended that you map out what your home (or 

small office) telephony requirements are. 

Compare the requirements with what facility you already have. You may find that you 

may need to provide extra facilities that you do not already have e.g. Network switches, 

router, ATAs or IP Phones and needless to say a suitable broadband service. 

For the purpose of this guide, I have determined that we will start with the following: 

• 2 extensions with voicemail using a Linksys PAP2 ATA 

• 1 extension using a Linksys SPA3000 ATA 

• 2 extension using Softphone (we will use these 2 extension for initial testing) 

• 3 extension using IP Phones 

• 2 VoIP service providers 

• 2 PSTN trunks – for the purpose of this exercise, I will be using a Linksys 

SPA3000 for one PSTN trunk and a X100P for the other PSTN trunk. 

• IVR 

• Music on hold 

• A number of customised voice prompts and announcements 

Once the above are set-up, I will customise it further with the following; 

• Time conditions 

• Ring groups 

• Follow me 

• Conference facility 

• DISA 

• Etc. 

In this exercise we will cover all of the above (not necessarily in the stated sequence) and 

we may also look at other possibilities as we go along, if time and/or resources permit. 

 

This is not an industrial grade installation. This installation is for training 

purposes only and to assist new hopefull to get started with Trixbox and nothing 

more. 

Once you are clear on what you want, you can start the installation in earnest. 


3 INSTALLATION 

A PC is a prerequisite. You can either install Trixbox in a virtual machine in your 

Windows based PC (there is a version of Trixbox for this) or install Trixbox on a dedicated 

PC. The later is recommended given that once it is set-up, we will need to leave the PC 

running and not burden it with other daily applications such as playing games, browsing 

the internet or doing the children school assignments. 

For stability, I will suggest that Trixbox be installed on a dedicated machine. This is what 

we are setting out to do. 

3.1 INITIAL INSTALLATION 

The Trixbox version that we will be installing is the latest current version available today – 

Trixbox 2.0. You will need to download the ISO from the Trixbox website below. 

http://www.trixbox.org/modules/smartsection/item.phpitemid=2/ 

After you have completed the download, 

• Burn the ISO image that you have just downloaded to a blank CD. 

• Ensure that your PC will boot from the CD. If necessary change the BIOS 

settings to enabled this. 

• **NOTE: This will erase all data on the hard drives of the PC. 

If you have 2 drives, both may be blown away as well - beware. 

• Boot your Trixbox PC with the CD in the CD Drive and press enter. After a few 

seconds, the following screen will be displayed. You press F2 to see the various 

options. However, it is not really necessary. Just press [Enter] to start the 

installation 

• Press the Enter key to start the installation. 


• After initial system detection, you will be asked to select the type of keyboard that 

you will be using. Use your cursor keys to navigate the keyboard types and use 

the Tab key to move the focus to the OK or Back option. Select the appropriate 

keyboard. Selected us and move the focus to OK and hit [Enter] 

• After system hardware probing is completed, you will be asked to select the Time 

Zone you are in. 

• Use your cursor arrow keys to navigate to the appropriate time zone. In our case, 

we will selected Australia/NSW. Use the TAB key to move the focus to OK and 

hit [Enter] to accept (see cheat sheet to set NTP time manually). 

• Next you will be asked to enter your root password (remember this password). 


• After entering and confirming your password, installation will now commence by 

first formatting the Hard Disk/s. 

• From this point it will take about 30-45 minutes for the installation to be complete 

and ready for the configuration stage. 

• During this stage, you will see screens similar to the following. Linux and the 

required files are being installed. All you do is, wait for it to finish. 

• After Linux is loaded the CD will eject. Take the CD out and wait for the system to 

reboot. 

• When it reboots, you will be presented with the following screen. 


• After a moment, it will continue and you will see lots of lines of codes. This 

process will take a while because it is building Asterisk. 

• When Asterisk build is complete, it will reboot itself. Once rebooted, Trixbox is 

ready for you to continue configuring and make changes to the system default. 

3.2 CHANGE DEFAULT SETTINGS 

Once Trixbox has been installed, some changes need to be made to secure your 

installation. 

Log in to your new Trixbox (user: root, password: The one you gave earlier) 

 

Trixbox 2.0 uses freePBX 2.1.3. To take full advantage of the latest freePBX 

features, lets upgrade the existing freePBX to the latest version 2.2. See the section 

on upgrading freePBX. Once that is done we will come back here. 


3.2.1 To get Help 

At the command line, type 

help-trixbox 

A list of help will be displayed – see illustration below: 

 

The help screen will also tell you the necessary command required to 

change the password of the various default users e.g. passwd-maint for 

user maint and passwd for user root. 

3.2.2 Change IP Address (set IP address to Static) 

(If you are installing Trixbox v2.6.0.5, I suggest that you DO NOT use NetConfig, instead 

use webmin or the Trixbox GUI to change your IP address because using NetConfig may 

cause you some problem) 

The following example is for those still using Trixbox v2.2.x or earlier on Centos prior 

to version 5. For Centos 5, please refer to the second part of this section. 

To change Asterisk IP address from DHCP to Static. At the command prompt enter: 

Netconfig 

You will see the following: 


Select [Yes] to set up networking and hit enter. 

You will then see the following screen. 

(If you are installing Trixbox v2.6.0.5, I suggest that you DO NOT use NetConfig, instead 

use webmin or the Trixbox GUI to change your IP address because using NetConfig may 

cause you some problem) 

Use the Tab key to cycle through the fields. Enter the IP address that is to be 

allocated to the Asterisk box, the Netmask (subnet mask), Default Gateway, and 

Primary nameserve as per the example above. In my example above, I used my 

existing network regime. 

• In the IP address field, enter an IP address Trixbox making sure it is within your 

network range e.g.: 192.168.1.100 

• Netmask is normally 255.255.255.0 unless your network has different Network 

mask. 

• Default gateway IP is the address of your router. In my case, my router address 

is 192.168.1.1 

• Primary nameserver is the address of your Name Server, usually your domain 

server if you have a network domainserver but if you are running workgroup, use 

the address of your default gateway e.g.: 192.168.1.1 

Once done, select OK. 


Those using Centos 5 are advised to use the Trixbox GUI to change the IP address 

to a Static IP address (see illustration below). 

Hover mouse on the System tab and select Network. 

Select the appropriate interface device as per the illustration below. 

Change the IP address appropriately. 

Those are the initial inputs that require immediate attention. Others, such as Admin 

password etc, are optional although it is recommended that you change them. Once 

done, log off Linux and reboot. 

amportal stop 

Shutdown –r now 

Asterisk will now start with the new IP address. 


4 SET-UP TRIXBOX 

Now, using your browser, you can connect to http://ipaddress/ (e.g. http:192.168.1.100) 

to configure Trixbox . 

You will be presented with the Trixbox initial User Mode splash screen as illustrated 

below. 

This screen (once Trixbox has been properly set-up) enables users to monitors ARI, FOP 

(flash operator panel), Client Manager (SugarCRM) and setup personal MeetMe 

requirement. 

As it stands, Trixbox need to be configured before any of those facilities are operable by 

users. 

Initial Welcome Screen (User Mode) v2.2 – 2.6 

To start configuring Trixbox, it is necessary to switch to Admin Mode. To do this, click on 

the “switch” label situated on the top right hand corner of the screen (marked with an 

arrow in the illustration above). 

A username and password dialog screen will appear where the default username and 

password need to be entered as per the illustration below. 

The Username is maint and the default password is password, unless you have 

changed the default password for user maint to something else using the passwd-maint 

command in chapter 3.2 above. 


Log in to System Administration 

 

To log in to System Administration, use user: maint, password: password unless you 

have changed the user maint password during initial set up in 3.2 

Once you logged in, you will be presented with the following screen 

Admin Mode Initial Configuration Screen 

4.1 INSTALLING TRIXBOX MODULES 

In most cases, all the modules that you will required have already been installed as 

defaulty. However, should you need other modules that were not included, you may need 

to install them from the online repository. At this stage, it is unlikely that we will need 

other modules. 

To install the Trixbox modules, you will need to log on to the Package Manager site to 

enable you to get a list of all the available modules (see Unable to log to Packages). 

I am already registered to the site and therefore I simply need to enter my username and 

password to login otherwise registration will be required before one can log on to the 

Package Manager site. 

If you are new, click on the “Register” button and register. Simply follow the prompt. 

However if you are unable to register through this method (some Firefox users are having 

this problem – Ed Note. Since the new interface, neither Firefox nor IE is able to register 

this way), you will need to go to http://www.trixbox.org and register to be a member of the 

forum. 


It is this username and password that you that is required for you to successully log in to 

Packages database (Please don’t ask me why this is so). 

Once logged in, select/click on the “Packages” tab (marked with an arrow above) and the 

current package list will be downloaded as illustrated below. 

Most of the modules necessary to get Trixbox operational are already pre-installed and 

ready for the individual administrator to start setting-up. However there are other optional 

modules that can be installed if necessary, MySqlManager, tbm-phpmyadmin.noarch is 

one such example. Without this package you will not be able to get to phpadmin through 

the GUI. 

For this installation, we will simply use the standard default modules and for the purpose 

of the exercise, we will additionally install “webmin” using the module manager. 

To find the package, you may need to scroll down the list till you get to the package called 

“webmin noarch” adjacent to Trixbox update as per the illustration below and place a tick 

in the install box (in the illustration it was shown that the tick was in the update box 

because webmin was already installed and to upgrade to a new version, place a tick in 

the upgrade box instead). 


Click the install or the upgrade button and webmin will be installed or upgraded. 

 

You will now have to wait for the installation to finish, as there is no progress 

indicator during the process. Patience is a virtue. 

How long you will need to wait depends on the number of modules that you selected. If 

you only select one module, it may only take a minute or so. If you selected more 

modules it will take much longer. Once done, we can focus on the rest of the chores. 

You are now ready to take the next step of installing freePBX modules as without these 

you will not have the GUI interface for managing Trixbox. There are other GUI interface 

that can be used but freePBX comes with Trixbox. 

 

Before you proceed further, it is a good idea to upgrade to the latest freePBX because 

by the time you install Trixbox, there is a good chance that there is a new version of 

freePBX. Besides, Trixbox did not come with the latest version of freePBX. 

To upgrade freePBX, go to the chapter relating to Upgrading freePBX. 

4.2 INSTALLING FREEPBX MODULES 

FreePBX is the GUI interface that Trixbox uses to manage the IP PBX. 

Before you can use freePBX you need to install all the modules that you require. You 

need to do this through freePBX. 

On the Trixbox Admin Mode screen, hover the mouse pointer over the Asterisk tab and 

select FreePBX from the dropdown list as illustrated below (see arrow). 

The following screen will appear. 


We are now in freePBX and this is where most of your configurations will take place. 

Select the “Module Admin” selection on the left. 

The next screen will then be presented with the list of available freePBX modules. Some 

of them will not be installed or enabled. 

It is up to you to select whichever modules you require to be installed in freePBX. 

If the orange bar as displayed above appears, simply ignore it for now. 

Start selecting all the modules that you want to install or enabled. To make it simple, let's 

install all the modules to start with (you can always remove modules or choose to disable 

some of the modules that you do not need at a later stage). 

 

You don’t have to select all the modules. Only select the modules that you need. 

Click 'Process' at the bottom of the list of modules once you have selected all of the 

modules. 

On the confirmation page, click 'Confirm.' 


You should get a page telling us that all modules were installed successfully. Then click 

return to go back to the modules list page. 

At this stage, click on the orange bar at the top of the page that says 'Apply Configuration 

Changes’ To confirm your intention, click on it. 

At the next prompt, select “Continue with reload” on the window that pops up. This 

reloads the Asterisk configuration and the orange bar will then go away. 

 

You will need to click on the orange bar anytime you make changes 

Before you go any further, you will check for modules updates that may be available or 

you may wish to include other extra modules that have been released since the 

installation of freePBX, online. 


We will go back to the Module Admin screen and lick on the “Check for update online” 

label as illustrated below. 

If there are updates that you need, simply select the necessary updated modules 

indicated in the list, click “Download and upgrade” radio button. 

The process is similar to the the initial installing and enabling of modules in the 

preceeding paragapgs. 

 

You can also uninstall or disable any module using this modular approach.. 

On completion, an orange bar at the top of the page that says 'Apply Configuration 

Changes’ will appear. Now click on it to confirm your intention then click 'Continue with 

reload' on the window that pops up. This reloads the Asterisk configuration and the 

orange bar bar will then go away. 

You can now click on ‘Setup’ tab next to the “Tools” tab to get to the modules you have 

just activated and finally start to configure the Trixbox IP PBX. 


5 BASIC CONFIGURATION USING FREEPBX 

From now on, we will be configuring Trixbox using freePBX web GUI as this is where 

most of the actions will be. 

5.1 GENERAL SETTINGS 

The first thing you will need to do is select General Setting and set it up as illustrated 

below. It is self-explanatory so I will not try to explain, as it is quite minimal and nothing 

substantial that warrants explanation. 

Notice that the extra information that will define the way Asterisk behave, are also 

required in freePBX. Set the fields to the following (these are vital settings): 

Asterisk Outbound Dial command option: 

“r” which generate the ring when you dial out, or “m” if you want music instead. There 

are other options of course – refer to the sub-chapter on Dial Command Options. 


Voicemail: 

Change the Direct Dial to Voicemail message type to something other than default - this 

will save you a lot of head scratching later on when you start using follow me etc. as your 

customs recorded message will not be played, instead the default Asterisk message will 

be played back if you don’t change the setting. 

I have set mine to Unavailable. 

Country Indications: Australia 

Allow Anonymous Inbound SIP Calls Yes (if this is set you ‘No’, all inbound unidentified 

SIP calls will not be accepted). 

Generally the default values are sufficient to get you started. These values can be 

changed later to suit your purposes. However the value I have as example above is 

required to ensure that Trixbox will function as expected – at the very list, during the 

testing stages. 

Naturally, help is available. Hovering your mouse on the corresponding field description 

with a dotted underline (e.g. Asterisk Outbound Dial Command options) will display the 

purpose of the fields and the various switches related to it. 

See illustration below. 

After setting up the General Settings, click on Submit Changes button and the red bar on 

top of the screen for the change to take effect. 

5.1.1 Dial Command Options 

In the Asterisk Dial command option, you may customise your preference to the 

way asterisk behave e.g. if you want the caller to hear music instead of the standard 

ringing sound, you may replace the “r” with an “m”. For further options, hover your 

mouse on the label and you will be informed of the other options. 

The following are the dial command options available to you but for now, lets leave it 

alone: 


Options: 

A(x) 

C 

d 

D([called][:calling]) 

f 

g 

G(contextêxten^pri) 

h 

H 

i 

L(x[:y][:z]) 

m([class]) 

M(x[ârg]) 

Play an announcement to the called party, using 'x' as 

the file 

Reset the CDR for this call 

Allow the calling user to dial a 1 digit extension while 

waiting for a call to be answered. Exit to that extension 

if it exists in the current context, or the context defined 

in the EXITCONTEXT variable, if it exists. 

Send the specified DTMF strings *after* the called party 

has answered, but before the call gets bridged. The 

'called' DTMF string is sent to the called party, and the 

'calling' DTMF string is sent to the calling party. Both 

parameters can be used alone. 

Force the callerid of the *calling* channel to be set as 

the extension associated with the channel using a 

dialplan 'hint'. For example, some PSTNs do not allow 

CallerID to be set to anything other than the number 

assigned to the caller. 

Proceed with dialplan execution at the current 

extension if the destination channel hangs up 

If the call is answered, transfer both parties to the 

specified priority. Optionally, an extension, or extension 

and context may be specified. Otherwise, the current 

extension is used. 

Allow the called party to hang up by sending the '*' 

DTMF digit. 

Allow the calling party to hang up by hitting the '*' 

DTMF digit. 

Jump to priority n+101 if all of the requested channels 

were busy. 

Limit the call to 'x' ms. Play a warning when 'y' ms are 

left. Repeat the warning every 'z' ms. The following 

special variables can be used with this option: 

• LIMIT_PLAYAUDIO_CALLER yes|no 

(default yes) - Play sounds to the caller. 

• LIMIT_PLAYAUDIO_CALLEE yes|no - 

Play sounds to the callee. 

• LIMIT_TIMEOUT_FILE File to play when 

time is up. 

• LIMIT_CONNECT_FILE File to play when 

call begins. 

• LIMIT_WARNING_FILE File to play as 

warning if 'y' is defined. The default is to 

say the time remaining. 

Provide hold music to the calling party until a 

requested channel answers. A specific MusicOnHold 

class can be specified. 

Execute the Macro for the *called* channel before 

connecting to the calling channel. Arguments can be. 

specified to the Macro using '^' as a delimiter. The 

Macro can set the variable MACRO_RESULT to 

specify the following actions after the Macro is finished 

executing 

• ABORT Hangup both legs of the call. 

• CONGESTION Behave as if line 

congestion was encountered. 

• BUSY Behave as if a busy signal was 


n 

N 

o 

p 

P([x]) 

r 

S(x) 

t 

T 

w 

W 

encountered. This will also have the 

application jump to priority n+101 if the 'j' 

option is set. 

• CONTINUE Hangup the called party and 

allow the calling party to continue dialplan 

execution at the next priority. 

• GOTO:^^ - 

Transfer the call to the specified priority. 

Optionally, an extension, or extension and 

priority can be specified. 

This option is a modifier for the screen/privacy mode. 

It specifies that no introductions are to be saved in the 

priv-callerintros directory. 

This option is a modifier for the screen/privacy mode. It 

specifies that if callerID is present, do not screen the 

call. 

Specify that the CallerID that was present on the 

*calling* channel be set as the CallerID on the *called* 

channel. This was the behavior of Asterisk 1.0 and 

earlier. 

This option enables screening mode. This is basically 

Privacy mode without memory. 

Enable privacy mode. Use 'x' as the family/key in the 

database if it is provided. The current extension is used 

if a database family/key is not specified. 

Indicate ringing to the calling party. Pass no audio to 

the calling party until the called channel has answered. 

Hang up the call after 'x' seconds *after* the called 

party has answered the call. 

Allow the called party to transfer the calling party by 

sending the DTMF sequence defined in features.conf. 

Allow the calling party to transfer the called party by 

sending the DTMF sequence defined in features.conf. 

Allow the called party to enable recording of the call by 

sending the DTMF sequence defined for one-touch 

recording in features.conf (default *1 in Asterisk v1.2) 

Allow the calling party to enable recording of the call by 

sending the DTMF sequence defined for one-touch 

recording in features.conf (default *1 in Asterisk v1.2) 


5.2 EXTENSIONS 

The number of extensions to be set up depends on you. You can have soft phones 

installed in 4 or 5 computers or mixture of ATAs and SIP SoftPhones. In this case, we 

have 8 extensions to be set up using ATA as well as softphone and IP phone as we have 

planned. 

It’s best to avoid the following extension numbers: 

 

200 - Park Notify 

300-399 - Reserved for speed dial (Backward compatible) 

666 - Reserved for FAX testing 

70-79 - Reserved for calls on hold (Definable) 

700-799 - Reserved for calls on hold (Definable) 

7777 - Reserved extension for incoming calls simulation 

5.2.1 Create Extensions 

To create extensions, do the following: 

1. Select the Setup Tab 

2. Click on Extensions 

3. Click on Add Extension 

4. From the drop down selection box, select Generic Sip Device since we are 

going to create a SIP extension 

5. Click Submit 

(see illustration below). 

Creating of SIP, IAX2, ZAP or Custom, can done from the Create Extension menu 

illustrated above: 

We need to create a few SIP extensons. Therefore you should select Generic SIP 

Device from the device drop-down list then click “Submit” as many times as you need 

to create extensions for. 

The illustration below is where you create the extension. 

You will notice a few fields that you will need to populate. The important ones are: 


• User Extension: 2000 (that’s the extension number I gave for reception) 

• Display Name: Reception 

• Secret: 2000 (for simplicity, I used the exten number as the secret password) 

I left the rest of the fields at their default values.. 

Click Submit when done. 

Add Extension Screen 

Click on the orange bar on the top of the screen after every time you have submitted 

a new extension. 

Click on the Add Extension button to add more extensions. 

We will create 8 extensions; 

2000 - Main Reception 

2001 - Medical Centre Manager 

2002 - Dr. Stan Roberts 

2003 - Dr. Susan Forbes 

2004 - Dr. Francis Wong 


2005 - Admin Office 

2006 - Lab and X-Ray 

2007 - Courtsy Phone. 

For simplicity, we will allocate passwords to be the same as the extension numbers. 

Where Voicemail is enabled, we will allocate the same password as well (but you 

don’t have to. You may also nominate an email address for Voicemail Email 

Notification – it’s up to you). Voicemail will be covered in more detail a little later in 

this document. 

 

If you want this extension to be a remote extension, you will need to edit the 

extension and ensure that the entry under Device options reads Nat: yes. 

5.2.2 Enable Voicemail 

To enable voicemail on an extension simply "enable" it when you create the 

extensions from the freePBX GUI. If you require email notification of your voicemail, 

you may enter your email address in the email address field. 

In this case let us enable voicemail on extension 2001 (the Manager’s extension). 

Go back to the Extension Option of freePBX and click on extension 2001. Scroll 

down to the Voicemail and directory section and do the following: 

• Status: Enabled (use the drop down selection to select it) 

• Voicemail password: 2001 (to keep it simple we shall use the extension 

number) 

• Email address: Enter an email address of the person (only if you want email 

notification) 

• Email attachment: yes (mark the radio button) 

• Play CID: yes 

• Play envelope: yes 

• Vm context: default 

Click submit when done followed by the red bar on top of the scree. 


5.3 FOLLOW ME 

After setting up the extensions, you need to decide if you want Asterisk to call another 

pre-arranged extension, if the extensions called do not answer. This is where we will 

define it as per the illustrations below: 

To do this, select the Follow me option; Setup -> Follow Me. 

You will be presented with the following screen: 

. 

Select the extensions that you want to define (the extension selection is on the right of the 

screen). In this case, we will pick the Medical Centre Manager’s extension – ‘Manager 

add’ 

In the screen that follows (see the illustration above), enter the following information. 


• Ring strategy: hunt (to call the numbers in sequence) 

• Extension List: 2001 (Manager’s extension) followed by Administration’s Office 

and the Reception (The reception will take down messages for the Manager). 

• Ring time: 40 seconds 

• Destination if no answer: Core – Hangup 

Every time the Manager’s extension is called, Asterisk will try to connect to extension 

2001 and if no answer, it will call Ext. 2005 followed by the Reception. If still no answer, it 

will simply hang up. During business hours, the reception will take down messages for 

the manager or the Doctors. 

Click “submit” followed by clicking the red bar on top of the screen to finalise this 

selection. 

5.4 RING GROUPS 

A ring group is a group of extensions that will ring when there is an external incoming call. 

You can even put your Mobile Phone number in the ring group if you want to. The 

0400123456# is the Medical Centre’s Manager phone (see illustration below). For mobile 

phone to work, you must have the appropriate route and trunk set up. 

You may not want a ring group – it’s entirely up to you. If you don’t require a ring group, you 

may ignore this section. 

When there is an incoming call to the ring group, the phones nominated in the selected 

group will ring. You may select different ring group for each of the incoming trunk or you 

may nominate the same group for all the trunks, in which case you will only need to define 

only one ring group. 

For simplicity, we will only defined 1 ring group for all incoming calls from all trunks – at 

this stage, let’s not get too fancy. 

We will create a ring group 600 for this purpose. Let’s call it AdminGroup as it rings all the 

phones related to the Administration of the Medical Centre. A single digit ring group is not 

recommended. 

We have included the Manager’s Mobile Phone number in as the last one in the group. If 

noone answer the call, Asterisk will simply hang up. Since the last number in the group is 

the Manager’s mobile phone, generally, the Mobile phone voicemail will take the message 

if the phon is not answered. 

The ring group screen is illustrated below: 


Now it’s a good time to set up your softphones. 


6 SET UP SOFTPHONES. 

There are a number of softphones available for use with PiaF. Some of them are free. 

The following are 2 examples of the free softphones. 

6.1 COUNTERPATH X-LITE SOFTPHONE 

X-Lite Softphones can be downloaded here: 

http://www.counterpath.com/xlitedownload.html 

After installing the softphone (assuming that users can install the softphone), we need to 

configure the sofphone. 

When you start X-Lite 3 for the first time, you will see the following screen. 

Click the Add button. 

You will then get the following screen. 

This is where you will enter your credential 

At the various fields, add the following: 

• Display Name: Your Name - not Ben 

Sharif ☺ 

• User Name: Your extension number 

• Password: The password of the 

extension when you created it in Elastix 

• Authorization User name: The same as 

your User Name or extension 

• Domain: Your Elastix IP address 

Put a check mark in the Register with domain 

and receive incoming calls 

Make sure the Target Domain Radio Button is 

marked. 

For this purpose, that’s all you need to do and Click apply, OK and close at the next 

screen. You are now ready to use the X-Lite softphone. 


Sometime it is necessary to change the Honor setting in X-Lite for it to work correctly. To 

do this, you would need to go to the advance setting of X-lite. 

Those using X-Lite having this problem may have to make the following correction in the 

advance setting of X-Lite. 

How to get to the advance setting 

In X-lite: 

Dial ***7469 (SEND) 

This will bring up the advanced settings window 

Filter for honor 

Double click on the honor entry and change the value to 1 

6.2 BOL SIPPHONE 

I have found the BOL SIPPhone extremely 

simple to set up for use with Asterisk and it 

also has a call forward facility that I use from 

time to time. 

You may obtain a copy of the BOL 2000 

sipphone from the link below. 

http://www.bol2000.com/website_c/download/sip 

phone/BOL%20SIPPhone_EN.msi 

After downloading and set up you will see the 

following when it is run: 

To configure the softphone, click on the hammer icon and you will see the following. 

Profile Tab 

This is the only screen that is required to be 

filled in. 

These are the only information required: 

Account: 

Password: 3001 (I use the same number as 

password for simplicity. I use the same 

password when I set up the extension in 

Asterisk). 

Domain/Realm: 

Proxy: Your Asterisk network address e.g. 192.168.1.100 

Port: 5060 

Check the Auto Login and Keep Password. 


Then click OK. 

Audio & Video Tab 

Next Click on the Audio & Video Tab to ensure 

that audio properties set is consistent with the 

Audio card installed in your PC/Notebook. 

The illustration above depicts the sound device installed in my Notebook. 

• Click on the Tuning Wizard to tune your audio input and output. 

• Check Auto Send Video (if you are using Video). I checked it anyway. 

• Check Auto Receive Video (if you are using Video). I checked it anyway. 

Click OK 

Network Tab 

Ensure that your Internet Connection Type is 

set to LAN. 

You may ignore the Information of Network 

field. 

Click OK 

STUN Tab 

You may or may not need to use a STUN 

server. If you need to use it, click on the 

STUN tab and enter the STUN server you 

want to use. If you do, a list of publicly 

available STUN server is listed in the section 

referring to STUN Servers later in this 

document. 

In my case, I do not use STUN and therefore I left that section blank and Enable STUN 

check box un-ticked. 

Click OK to close. 


Call Forward 

This is pretty simple to set-up. To forward an unanswered call to this extension, all 

you need to do is click on the Call Forward tab and enter the telephone number you 

want to forward your incoming calls to. You have 3 options of call forwarding – 

Always, On Busy or On No Answer. However this facility is only available if your PC 

is on and the softphone is active. 

Click OK when done. 

You might want to set-up a couple of PCs with the softphone after which you may start 

testing your brand new Phone System by dialling each extension in turn. 

If you use one of the softphone and dial 7777, Asterisk will simulate an incoming call. 

Once done, you may test your softphone connection to Asterisk. 

If everything has been done as explained above, we should be able to make and receive 

calls between our internal extensions. If not, it is time to re-inspect what we have done 

above and make the necessary correction before attempting to go any further. 

Let’s take a break and test the soft phone extensions by making calls to each extension 

that we have created. 


7 TRUNKS TO THE OUTSIDE WORLD 

Making and receiving calls between internal extensions is one thing, but a telephony 

installation isn’t really an installation if you can’t receive and make calls to the outside 

worlds. 

To be able to do these, you will need at least one trunk to the outside world. 

This is a simplified guide to give the budding Trixbox users some understanding in setting 

up Trunks. 

 

Points to keep in mind. 

Beware!!! If you are using an ATA, please allow for what the ATA will be sending 

to asterisk when you are preparing you rules and patterns. If you want asterisk to 

take care of all the rules, it is best to ensure that your ATA is sending exactly what 

you dialled to Asterisk – just delete your dialling rule from your ATA and replace it 

with (**|*x.|x.|**x.) 

7.1 WHAT IS A DIAL PATTERN 

Dial patterns are part of Outbound Routing. They act like a filter for matching numbers 

dialled with trunks. 

The various patterns you can enter are similar to Asterisk's definition of them: 

• X — Refers to any digit between 0 and 9 

• N — Refers to any digit between 2 and 9 

• Z — Any digit that is not zero. (E.g. 1 to 9) 

• [Various] — Match only one character that matches any of the one in the square 

brackets. (E.g. [02-68*#] would match 0, any number between 2 and 6 inclusive, 

8, * and #. Or, another way of saying this would be 'Match * or #, or a number that 

isn't 1,7 or 9') –Avoid using this notation in your Trunk Dial Rule as it does not seem 

to work. Use this only in your Outgoing Route Dial Pattern. 

• . — Wildcard. Match any number of anything. Must match *something*. 

• | — This lets you use a '0 to dial out' (or '9' in the US) by matching anything 

before the line, but not sending it to the trunk. 

7.2 WHAT IS A TRUNK 

A trunk is the telephony service line that you will be using to make an external call on, e.g. 

the Telstra telephone line that you have at home is a trunk (or it can be used as a trunk if 

you wish but not necessarily so). A VOIP service provider (VSP) that you have signed up 

with is also a trunk. 

If you have paid for VOIP service to enabled you to make calls out through PSTN, you 

can use this as a trunk for that purpose otherwise, you can only use it for making calls 

using VOIP between subscribers of the VSP only (unless there are peering arrangements 

with other VSPs). 

You can have several trunks if you want to. You can have Telstra and/or Optus 

telephone lines and use them as trunks, and you can also have a few VSPs that you 

subscribed to as additional trunks. 

To make external, PSTN or VOIP calls; you must have at least one trunk. 


The reason why asterisk users have several trunks can be explained as follows: 

• Trunk (a) charges the best rate for Local calls but is expensive for Mobile calls and not so 

great for international calls. 

• Trunk (b) has great rate for international call. 

• Trunk (c) has good mobile rate and will allow 1300 numbers while the other 2 do not. 

A properly planned route will direct the phone calls you make to the appropriate trunk that 

will provide you with maximum effectiveness and savings. 

E.g. When you make a call to a Mobile phone, asterisk will route your call via trunk (c) 

while it will route your call to trunk (a) if you make a local call. Similarly, when you make 

an international call, asterisk will route your outbound call via trunk (b). 

7.3 TRUNKS AND OUTBOUND ROUTES – A CIRCUS OF CONFUSION 

To the uninitiated, Outbound Route Dial Patterns and Trunk Outgoing Dial Rules are 

2 areas of confusion. I will try to explain this in the simplest possible term. 

Each of these two works in conjunction with each other – they compliment each other 

when an outgoing call is being made. 

7.3.1 Outgoing Route Dial Pattern 

This is the pattern you normally will dial when you want to make a phone call. 

As an example, if you are living in Brisbane, and you are calling a friend who lives in 

Sydney, you will normally dial a number such as 02 91234567, however if that friend 

is in Victoria, you will need to dial 03 91234567. 

Since your VOIP provider does not discriminate where you are calling from or where 

you are calling to, it requires a universal type number to call so your call can be 

directed to the correct recipient. 

In most cases, the VOIP provider will need to dial the universal e164 format e.g. 

61291234567 for Sydney or 61391234567 for Melbourne. (61 being the code for 

Australia, 2 or 3 being the code for Sydney or Melbourne respectively, followed by 

the telephone number of your friend which is 91234567). 

Since you are dialling 03 91234567, asterisk only need to send 391234567 (without 

the 0 prefix). Therefore in your Outbound Route Dial Pattern (lets call this Outgoing 

Route Domestic) you will need the following pattern: 

0|NXXXXXXXX 

That pattern tells Asterisk to send only 9 digits out of 10 dialled by not sending the 

first digit 0. Thus, if you dialled 0291234567, the number that will be sent to the 

Trunk will be 291234567. 

How will this then be translated to 61291234567 This is where the Trunk Outgoing 

Dial Rule comes in. 

7.3.2 Trunk Outgoing Dial Rules 

This is what the trunk is required to send to the VSP to make a successful call to the 

number you dialled. 

Using the above example, assuming that the VOIP service provider requires the full 

e164 format, Asterisk will need to add the missing Australian code 61. 


In your Trunk Outgoing Dial Rule, you will need the following pattern. 

61+NXXXXXXXX 

That pattern tells Asterisk to add 61 in front of the 291234567 before dialling the 

number via the trunk. For variations of patterns, see What is a Dial Pattern in the 

previous chapter earlier on in this document. 

7.4 LET’S CREATE THE VOIP TRUNKS 

To create a new trunk using freePBX, select Setup tab and then select the Trunks option 

from the vertical menu on the left. 

For this example, let’s create 2 SIP trunks; Pennytel and Oztell. 

Pennytel will be used for all International and mobile calls 

Oztell will be used for all local and mobile calls. 

To create a new Trunk, click on the Add Trunk option (first option on the right of the 

screen). 

General Setting (This applies to all the trunks): 

Outbound Caller ID: “Insert Your Name” (Must include the “ “ and ) 

Maximum channels: Leave blank for maximum. 

In this example we will create a Pennytel and Oztell SIP Trunks, which we will call 

pennytel and oztell respectively. We will also define their dialling patterns. 

7.4.1 Pennytel 

In this example, we will use Pennytel for all international and mobile calls only. 

Pennytel requires numbers dialled to follow the international format Country code + 

area code + phone number. To call an overseas destination you would normally dial 

0011+country code + phone number. E.g. to call New Zealand is 0011 64 , and to dial France is 0011 33 . Since Pennytel only requires 

the country code and the phone number but does not require the 0011, the route (that 

we will create later) will not pass the 0011 to the trunk. 

We will require the following Outgoing Dialling Rules for this trunk, which instruct the 

trunk to dial whatever is sent by the appropriate route. This example is for dialling 

Malaysia with a country code of 60, and Australian Mobile phones. 

Pattern 

601XXXXXXXX 

60NXXXXXXX 

61+4XXXXXXXX 

Explanation 

This pattern tells Pennytel to dial all Malaysian Mobile numbers 

starts with 601 followed by 8 more digits. 

This pattern dials numbers start with 60 followed by a one-digit 

number in the range of 2-9 followed by 7 more digits. 

For dialling Australian Mobile phone. The 61 is the Australian 

Country code that will be added to the mobile phone number. 

In your Outgoing Dial Rules box, we will enter the 

following; 

601XXXXXXXX 

60NXXXXXXX 

61+4XXXXXXXX 

Next we need to create the Outgoing Setting, Incoming Settings and Registration for 

Pennytel. 


Outgoing Settings 

In the Trunk Name field we will enter the name of this trunk: pennytel 

In the Peer Details enter the following; 

Pennytel preferred codecs are: g729, gsm 

and g723 but g729 and g723 are not installed 

as default in TRIXBOX 

allow=alaw&ulaw&gsm 

canredirect=no 

canreinvite=no 

disallow=all 

host=sip.pennytel.com 

insecure=very 

secret= Password 

type=peer 

username=8880XXXX 

Incoming Settings 

In the User Context, enter the Pennytel User ID: 888XXXXX 

In the User Details we will have the following: 


context=from-trunk 

fromuser=888XXXXX 

qualify=no 

secret= Password 

type=user 

username=888XXXXX 

Registration 

In the registration field, enter the following: 

888XXXXX:Password@sip.pennytel.com/888XXXXX 

After all these have been done, click on the Submit Changes button and click on the 

red bar across the top of the screen. 

 

It seems that Pennytel is rejecting Asterisk. To overcome this problem, it is 

necessary to add the following in the general section of your sip.conf. 

useragent = Trixbox 

The useragent can be any name, as long as it is NOT Asterisk. 

My sip.conf looks like this; 

; Note: If your SIP devices are behind a NAT and your Asterisk 

; server isn't, try adding "nat=1" to each peer definition to 

; solve translation problems. 

[general] 

port = 5060 ; Port to bind to (SIP is 5060) 

bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) 

disallow=all 

allow=ulaw 

allow=alaw 

; If you need to answer unauthenticated calls, you should change this next line to 


; 'from-trunk', rather than 'from-sip-external'. You'll know this is happening if when 

; you call in you get a message saying "The number you have dialled is not in service. 

; Please check the number and try again." 

context = from-sip-external ; Send unknown SIP callers to this context 

callerid = Unknown 

tos=0x68 

useragent = Trixbox 

; #, in this configuration file, is NOT A COMMENT. This is exactly how it should be. 

#include sip_nat.conf 

#include sip_custom.conf 

#include sip_additional.conf 

7.4.2 Oztell (SIP) 

Oztell requires the numbers dialled to follow international format Country code + area 

code + phone number. To call Sydney 67541234, Oztell requires it in the following 

format 61 267541234. To use Oztell family cheap rate you can also prefix it with 09. 

Therefore the dialling rule will be in this form: 

We need various patterns here: 

Pattern 

Explanation 

899060X. This pattern tells Oztell to dial the inter Oztell’s numbers 

This pattern dials numbers start with 60 followed by a one 

digit number 1-9 followed by other numbers minimum of 2 

60ZXX. 

digits from 0 -9. This can be an international number with a 

Country code of 60 followed by a number between 1 and 9 

inclusive. 

06612+NXXXXXXX Using the Oztell 06 rate to dial any NSW number. 

0661+NXXXXXXXX Using Oztell 06 rate to dial any Australian number. 

61+1300XXXXXX Dials 1300 numbers 

61+13ZXXX Dials 13 numbers 

61+1800XXXXXX Dials 1800 numbers 

In your Outgoing Dial Rules box, it will look like this: 

– this is simply an example based on my setting; 

yours will no doubt be different. 

899060X. 

60ZXX. 

06612+NXXXXXXX 

0661+NXXXXXXXX 

61+1300XXXXXX 

61+13ZXXX 

61+1800XXXXXX 

Next we need to create the Outgoing Setting, Incoming Settings and Registration for 

Oztell. 


In the Trunk Name, enter the name of this trunk: oztellsip (as an example) 

In the Peer Details we will need the 

following; 

You must enable all the codecs that you want to 

use through Oztell’s web page. 

allow=ulaw&alaw 


disallow=all 

dtmfmode=rfc2833 

fromuser=899060xxxxx 

host= sip2.syd.net2max.com 

insecure=very 

secret=Password 

type=peer 




In the User Context, enter the Oztell User ID: 899060XXXXX 

In the User Details enter the following: 

Registration 



fromuser=899060XXXXX 

insecure=very 

qualify=no 


type=user 



899060XXXXX:Password@sip2.syd.net2max.com/899060XXXXX 

After all have been done, click on the Submit Changes button and click on the red 

bar across the top of the screen. 

6.4.3 Oztell (IAX) 

Since I have IAX trunk with Oztell, I might as well include my Oztell IAX trunk setting 

here for general information only just in case you also want to set up an IAX trunk. 

Note* Don’t forget to Forward UDP Port 4569 to your Asterisk IP address and edit your 

iax.conf file. 


In the Trunk Name, enter the name of this trunk: oztelliax (as an example) 

In the Peer Details we will need the 

following; 

Note: You must enable all the codecs that you 

want to use through Oztell’s web page. 


disallow=all 

host=iax2.syd.net2max.com 

qualify=yes 


type=peer 



In the User Context, enter the Oztell User ID: 899060XXXXX 

In the User Details enter the following: 



type=user 

Registration 


899060XXXXX:Password@iax2.syd.net2max.com 

After all have been done, click on the Submit Changes button and click on the red 

bar across the top of the screen. 


The SIP Trunks screen in freePBX will look like the following illustration below: 

* For more trunk settings, please refer to APPENDIX B 

Now that you know what the trunk will do, you may proceed to the next chapters to 

create the Inbound and outbound routes. 


8 CALLS HANDLING 

8.1 INBOUND ROUTES (INCOMING CALLS) 

This is where the behaviour of incoming calls from all trunks is being handled. 

When an incoming call from PSTN or VoIP trunk is received, asterisk needs to know 

where to direct it. It can be directed to a ring group, an extension, Digital Receptionist 

(IVR) or Queue. For this purpose, Inbound Route needs to be set up. 

Select the Inbound Routes selection in the left bar of the screen. 

Initially, (since we have not created an IVR yet) we will set all incoming calls to be 

directed to Ring Group 600 as illustrated below (later on in this guide we will create a 

digital receptionist for this purpose and we will redirect all the incoming calls to the IVR). 

 

As a matter of fact, this can be the only Inbound Route that you will need if 

you deploy IVR (to be discussed later on). Besides, without this inbound 

route created, your 7777 simulated calls will not work. 

For our purpose, only put in the following data. The rest are optional. 

• DID Number: Leave blank 

• CID Number: Leave blank 


• Fax Extension: Disabled (I do not require fax facility here) 

• Privacy Manager: No 

• In the radio button, enable Ring Group 600 (AdminGroup). We want all 

incoming calls to ring the phones in the Admin group (see illustration below). 

For each extra trunk, a corresponding inbound route may be created (you don’t have to). 

All incoming calls from all the different trunks that you may have can have their own 

individual characteristic by judicious deployment of the time condition facility available in 

FreePBX (to be discussed later on). 

However, at least one Inbound Route must be created for Trixbox to answer incoming 

calls. This includes calls from PSTN trunk as well. The route that we have created above 

is precicely for this purpose. 

This route will handle the calls that come in without Trunk ID and calls from trunks that do 

not have an Inbound Route created (which includes all calls from PSTN and incorrectly 

created incoming route). We call this a “Catch-All” route, an Inbound Route with the DID 

Number and Calling ID Number fields left blank. 

As usual, you will need to click on the red bar on top of the screen after each time you 

submit a new Inbound Route. 

8.1.1 Inbound Route from ZAP Trunk 

To handle inbound calls from a ZAP trunk, enter the Zaptel Channel number in the 

zaptel channel field. This will determine which zap call be directed to where. If there 

are 4 FXO modles in a TDM400 card, the zap channels will be 1, 2, 3 and 4. Each 

FXO is a channel (this can be confirmed in zapata_auto.conf file). 


8.2 TIME CONDITIONDS 

You can create various time conditions and use these time conditions in conjunction with 

your Inbound Route to individualise each of the incoming trunk’s behaviour. 


You may create several time conditions and give each of the time condition a Short Name 

to identify it. These time conditions can then be assigned to each individual Inbound 

Route if you choose to do so or they can be nested. 

8.2.1 Simple Time Condition 

Unlike businesses, most private householders, ignore the time condition, instead 

would prefer to direct all calls to an extension or selected extensions regardless 

whether it is day or night. If this is the case, you may ignore time condition 

altogether, instead, use the inbound route to direct your incoming calls to individual 

extensions or to the digital receptionist. However, if you want to accept calls during 

the day but prefer callers to simply leave voice messages during the night, below is 

an example of a simple time condition that you can adopt where the time condition 

will send the calls to IVR during the hour of 8:00am and 8:00pm and anything outside 

the stipulated hours, the calls will be directed to your voicemail. 

• Add a new time condition by selecting Setup - > Time Condition - > Add 

time condition. 

• In the Time condition name field, give it a name eg: DayNight 

• Under Time to match, do the following: 

o Time to start: 08:00 

o Time to finish: 20:00 

o Week Day start: Monday 

o Week Day finish: Sunday 

• Under Destination if time matches, select IVR (or Core extension or ring 

group) 

• Under Destination if time does not match, select Core mailbox: eg 

voicemail box 2001 (if you have voicemail enabled for ext. 2001). 

• Submit and click on the red bar on top of the page. 

Once the above is done, go to Inbound Route, select the appropriate inbound route 

and under Set Destination, select Time Condition DayNight, submit and click the red 

bar on top of the screen. 

8.2.2 Complex Time Condition 

A slightly more complex one can also be crafted e.g: You want o accept calls only 

between 8:00am till 1:00pm and between 2:00pm and 5:30pm Monday to Friday and 

any other time you want to direct all the calls to voicemail. 

1. Add a morning time condition. 



• In the Time condition name field, give it a name eg: OfficeMorning 





o Week Day finish: Friday 


group) 

• Under Destination if time does not match, select a core extension for the 

moment. We will need to change this later. 


2. Add an afternoon time condition 



• In the Time condition name field, give it a name eg: OfficeAfternoon 






o Week Day finish: Friday 


group) 

• Under Destination if time does not match, select Core mailbox: eg 

voicemail box 2001 (if you have voicemail enabled for ext. 2001). 


3. Go back to time condition OfficeMorning and change the Destination if time 

does not match to: Time Condition OfficeAfternoon. Submit and click on the 

red bar on top of the screen. 

4. Once the above is done, go to Inbound Route, select the appropriate inbound 

route and under Set Destination, select Time Condition OfficeMorning, submit 

and click the red bar on top of the screen. 

What the inbound route will do is, as soon as an incoming call is accepted, it will be 

forwarded to the OfficeMorning time condition. 

If the time condition is met, it will then be forwarded to the IVR. On the other hand, if 

it does not match, it will be forwarded to time condition OfficeAfternoon. 

If it matches the OfficeAfternoon time condition, the call will be forwarded to IVR, if 

not, it will be forwarded to the voicemail box that you have determined. 

Following the above example, time conditions can be nested to multiple levels deep 

allowing you to manage your time condition and make them as complex or as simple as 

the situation demands – imagination is the only restricting factor. 

8.3 OUTBOUND ROUTES 

8.3.1 What is an Outbound Route 

An outbound route works like a traffic cop giving directions to road users to use a 

predefined route to reach a predefined destination. 

8.3.2 How does it work 

Every time you dial a number, asterisk will do the following in strict order: 

• Examine the number you dialled. 

• Compare the number with the pattern that you have defined in your route 1 

and if matches, it will initiate the call using that trunk. If it does not match, if 

will compare the number with the pattern you have defined with route 2 and 

so on. 

• Pass the number to the appropriate trunk to make the call. 

 

To make an external call (except inter extension calls), you will need at least 

one trunk and one route. 

In this case we will create 5 routes: 

• International 

• Domestic 

• MobileAust 


• OztellOnly 

‣ The International Route will be used for all international phone calls. 

‣ The Domestic Route will take care of the Australian phone and mobile. 

‣ The other two will do as the names suggest. 

To create a new route using freePBX, select Setup tab and then select the 

Outbound Route option from the vertical menu on the left. 

Click on the Add Route in the menu on the right of the screen. 

8.3.3 International 

You will need the following pattern to match international dialing. 

Pattern 

0011|60NXX. 

0011|[7-9]XX. 

0011|NXX. 

Explanation 

Most of my International calls are to Malaysia, I have defined 

the Malaysian PSTN pattern here. 

This pattern dials all international numbers by prefixing it with 

0011 and any digit between 7 and nine inclusive. Before the 

number is sent to the trunk the 0011 will be stripped. If 

required by the trunk, the trunk will insert the 0011. 

This is the catchall international dialling including International 

Mobile (excluding USA which is a 1). 


In your Dial Patterns box, you will see something like this. 

Dial Patterns 

0011|60NXX. 

0011|[7-9]XX. 

0011|NXX. 

You will then need to pick the Trunk Sequence. In this case we will use Pennytel as 

the trunk for this route and if it fails, we will use Oztell as a second choice/backup. 

Trunk Sequence 

0 Pennytel 

1 Oztell 

Any number dialled with the 0011 prefix will be routed through the Astratel trunk and 

should it fail, it will be routed to Pennytel. 

8.3.4 Domestic 

You need the following pattern to match the Australian Telephone numbers and the 

Australian Mobile numbers. 

Pattern 

0|[23478]XXXXXXXX 

[2-9]XXXXXXX 

ZXX. 

Explanation 

You will dial Australian telephone number as normal 

using 03, 02 prefix but the route will strip the leading 

0 before it is sent to the trunk. 

If the trunk requires the 0, the trunk will insert it. 

If you dial local number and not use the STD prefix, it 

will send to the trunk as is and the trunk will insert the 

02 if required (this is a NSW pattern. If you are in 

Victoria you will need to modify the trunk). 

This is the catchall dialling for numbers that does not 

start with a 0 such as 1300, 1800, 13 etc. 

In your Dial Patterns box, you will see something like this. 

Dial Patterns 

0|[2378]XXXXXXXX 

[4689]XXXXXXX 

ZXX. 

You will then need to pick the Trunk Sequence. In this case we need Oztell as the 

trunk for this route and if it fails, we want Pennytel to take over as a second 

choice/backup. 


0 Oztell 

1 Pennytel 

All numbers dialled matching the above pattern will be routed through Oztell and if it 

failed, it will be routed via Pennytel. 


8.3.5 MobileAust 

There is only a single pattern for calling Australian mobile – illustrated below. 

Dial Patterns 

0|4XXXXXXXX 

We want all my domestic mobile calls to be routed via Pennytel and we have not 

nominated a second trunk. If it fails then we will not be able to make a Mobile call, as 

we don’t have a second trunk nominated. 


0 Pennytel 

8.3.6 Oztellonly 

The following pattern is in the Oztellonly route. We will use Oztell for a number of call 

types therefore some explanation may be in order. 

Pattern 

Explanation 

This pattern will allow me to dial a Malaysian Mobile number. 

The 0011 will be stripped before the number is sent to Oztell. 

0011|601XX. 

Oztell does not require the 0011. It only requires the Country 

code. 

You will dial Australian telephone number as normal using 

03, 02 prefix but the route will strip the leading 0 before it is 

0|[378]NXXXXXXX 

sent to the trunk. 

If the trunk requires the 0, the trunk will insert it. 

1[38]XX. 

This is for dialling 1300, 1800 and 13 numbers. 

899060XX. For dialling Oztell inter Oztell users 

NXXXXXXX Catchall for dialling a local number that doesn’t start with 1 

In the dial pattern box you will see something like the illustration below. 

Dial Patterns 

0011|601XX. 

0|[378]NXXXXXXX 

1[38]XX. 

NXXXXXXX 

Since the above pattern is for Oztell only, in my trunk sequence, Oztell was chosen to 

be the only trunk for any of these calls. 


0 Oztell 


8.4 ENUM 

ENUM is a way for us to use VOIP to establish calls between VOIP systems without using 

PSTN even though the numbers called are PSTN numbers provided that the numbers 

called are registered with ENUM. In short, ENUM offers you the ability to call other 

telephones directly over the Internet without involving a third party. 

I will not enter into any elaborate explanation how enum works, as there are already a 

number of articles describing this. In essence, your phone number and server information 

are stored in a special DNS zone (very similar to a reverse lookup zone). When an 

ENUM enabled Asterisk user calls your phone number, it will first attempt to find your 

phone number in the ENUM DNS zone. If it finds your phone number it will use the 

information to setup a call directly between his Asterisk server and yours, resorting to 

PSTN only if your number is not found. 

Note: ENUM only works between ENUM enabled Asterisk (or other ENUM enabled VOIP) 

users. 

8.4.1 Setting up ENUM Trunk 

1. Click the Add Enum Trunk link. 

2. In the Outbound Caller ID field enter your callerid information in the format of 

"Your Name" (Including the quotes), assuming your telephone 

number in Sydney is 0246123456. 

3. On the outgoing Dial Rule enter 612+NXXXXXXX (for local calls without having to 

dial the 02) 

4. On the second line enter 61+NXXXXXXXX (for dialling Australian numbers where 

you need to dial the prefix (eg. 03). 

5. On the next line enter ZXXXXXXXX. (including the dot) for dialling International 

numbers – see example below: 

6. Click the Submit Changes button at the bottom of the page 

7. Click the Apply Configuration Change at the top of the page. 


8.4.2 Setting up ENUM Outbound Route 

1. Click the Outbound Routes link on the left hand side. 

2. To create a new route, click on Add Route – the top option on the right side of the 

screen. 

3. Name the route eNUM. 

4. We will to use the prefix 164 to force dial through ENUM 

5. In the Dial Pattern, on the first line, we insert 1640|[378]NXX. This will allow us to 

dial an STD number such as 0732123456 by dialing 1640732123456 where only 

the 732123456 will be sent to the ENUM trunk. 

6. On the second line we insert 1640011|NXX. This will allow us to call an 

International number eg. To call Malaysia we will dial 16400116062861234 and 

only the 6062861234 will be sent to the ENUM trunk. 

7. On the third line I have 164|NXX. This enables me to dial NSW local number 

without having to dial 02. 

8. Make the ENUM trunk the first route in your Trunk Sequence. You may also 

add other trunks in the sequence in case ENUM failed. 

9. Click the Submit Changes button at the bottom of the page. 

10. Click the Apply Configuration Change at the top of the page. 

11. You may try calling someone while monitoring Asterisk CLI and you should see 

the Enumlookup scroll by. 

Click here for a Step-by-Step instruction on how to set up an ENUM account. 

 

There is a bug with freePBX 2.2.1 where ENUM trunk will not save. See 

ENUM Trunk won’r install under Bugs Report. 


9 CUSTOM CONTEXTS MODULE 

The custom contexts module currently is a beta module that must be manually installed. 

First a brief description of what custom contexts does (version 0.3.3): 

Custom contexts can be used to allow limited access to dialplan applications. 

Features: 

• Allows for time restrictions on any dialplan access. 

• Allows for pattern matching to allow/deny. 

• Allows for failover destinations, and PIN protected failover. 

• This can be very useful for multi-tennant systems. 

• Inbound routing can be done using DID or zap channel routing, this module 

allows for selective outbound routing. 

• Phones can be placed in a restricted context allowing them only internal calls. 

More info: http://aussievoip.com.au/wiki/freePBX-CustomContexts 

DOWNLOADING AND UNPACKING MODULE. 

Login on the Elastix box manually as user root and do the following: 

: 

cd /var/www/html/admin/modules 

wget http://www.freepbx.org/trac/attachment/ticket/1447/customcontexts- 

0.3.3.tgzformat=raw 

mv customcontexts-0.3.3.tgzformat=raw customcontexts-0.3.3.tgz 

tar -xzvf customcontexts-0.3.3.tgz 

rm -f customcontexts-0.3.3.tgz 

INSTALLING CUSTOM CONTEXT MODULE 

On the Trixbox/freePBX web interface click on Admin Setup Modules Admin 

At the Modules Admin screen, select Custom Contexts, select Install, click on Process. 


Confirm you action by 

clicking on the confirm 

button 

Click on the orange bar to complete the installation. 

ADD AND CONFIGURE NEW CUSTOM CONTEXT 

Add New custom Context: Admin Setup Custom Context 

To edit the custom-context, 

only select: Allow or Deny 

each context. 

To control any outbound call, 

select the route and from thes 

selection, select Allow or 

Deny. 


APPLY A CUSTOM CONTEXT TO EXTENSION. 

To finish, you use the custom context on 

Edit Extensions, Device Options. 

Reference 

http://aussievoip.com.au/wiki/freePBX-CustomContexts 

http://www.elastix.org/index.phpoption=com_openwiki&Itemid=27&id=installation#first_time_access_to_the_web_interface 


10 SYSTEM RECORDINGS 

System recording is a facility available under freePBX to enable the recording of 

customised voice prompts etc. 

By now we should be able to navigate ourselves through the freePBX menu. To get to 

the system recordings, select the Setup tab and pick System Recordings as per the 

illustration below. 

Clicking on the System recording label as highlighted by the red arrow will open the 

following “System Recordings” screen. 

There are two methods that you can capture your recording: 


• By recording directly using your telephone 

• Uploading pre-recorded materials 

10.1 DIRECT RECORDING 

Enter the extension number of the mouthpiece that you will be using to record your 

message in the appropriate field. 

You will be making your recording using the telephone on this extension (or the 

microphone of the softphone). 

Click Go. 

Follow the prompt on the screen and dial *77 on your phone to do your recording now. 

In my message, I have asked one of the receptionists to record the following: 

“Hi, you have reached the Family Health Centre. Press 1 for Reception, 2 for the 

Centre’s Manager, 3 for Dr. Roberts, 4 for Dr. Forbes, 5 for Dr. Wong, 6 for the Admin 

Office, 7 for Lab and X-Ray or you may simply hold and your call will be answered by one 

of our available receptionists. If no one answers your call, please leave a message”. - 

Then hang up. 

Verify the recording by dialling *99. Go to step 1 again if you are not happy with the 

recording. Keep doing this till you are satisfied. 

Once you are happy with the recording, give the recording a descriptive name e.g. 

“MainMenu” so you will know what the recording is for. 

Click Save. 

10.2 UPLOAD PRE-RECORDED MATERIAL 


If you have a recording that you have prepared in a .wav format you can simply upload 

that recording by browsing you local hard drive, locating the file that you want to upload 

and upload it to Trixbox. 

Follow the prompt for the upload and once again when the upload is completed, give the 

file a descriptive name for you to know what that recording is for. 

The above recorded files either recorded manually or uploaded using the system 

recording facility will be stored in the /var/lib/asterisk/sounds/custom directory. 


11 DIGITAL RECEPTIONIST 

Like most organisation, where possible, we would like to automate the redirecting of all 

incoming calls. The digital receptionist is very handy for these sorts of things. Unless the 

calls are non-specific and will need the assistance of the receptionist (in our case also the 

operator), the system should allow callers to make the selection. 

11.1 SETTING UP DIGITAL RECEPTIONIST 

In the Setup tab, click on the IVR option and get the following screen. 

Click on the Add IVR option on the top right on the screen (see arrow) and the new 

Unnamed IVR configuration screen will be presented to you. 

We will then fill in the following fields: 


Name (Change Name): MainMenu (that’s what we shall called the main menu) 

Timeout: 10 (leave it at default) 

Enable Directory: Tick 

Directory Context: Leave it at Default 

Enable Direct Dial: Tick 

Announcement: MainMenu (That’s what we called the IVR recording that was made for this 

purpose) 

(see illustration below) 

Then proceed to enter the selection options that you require for this IVR when the caller 

presses the appropriate option. 

If you need more choice options in the IVR, click on the Increase Options button at the 

bottom of the screen. 

In this IVR, we have defined our requirement as outlined below: 

For number 1 - 7, we select Extension, and for 8, which we have changed to the letter “t”, 

we also selected Extension. 

For 1, we will select the Reception (Extension 2000). 

For 2, select 2001, which is the Manager’s extension. 

For 3, select Dr. Roberts (Extension 2002) 

For 4, select Dr. Forbes 

For 5, select Dr. Wong 

For 6, select Admin Office 

For 7, select Lab and X-Ray 

For 8, (which we shall change to “t”) select extension 2000, which is the reception. 


The last extension (which we have changed to the letter “t”) is to capture timeout, if the 

caller does not press any option or, in case the DTMF tone does not register, the call will 

be sent to the receptionist. 

Once finished, click Save and also click on the Apply Configuration Changesed on the 

top of the screen. 

Test it by dialling 7777, (Asterisk will simulate an incoming call) and you will hear your 

Digital receptionist in action. 


11.2 MULTI-LANGUAGE IVR 

(This is included in this section as an example only and is not part of this tutorial. However you 

can use this if you want to. Besides, the latest FreePBX already has options for multi language 

implimentation.) 

Trixbox allows multi languages IVR handling by simply telling Trixbox what language to 

use. Naturally you will need to install all the language sets that you wish to use. 

See Installing Other Languages in the chapter referring to Customised Voice in later part 

of this document. 

I am assuming that you already have your MainMenu IVR in English set up for selecting 

various options. This IVR is known as IVR-2 in your extension_additional.conf file. Any 

subsequent IVR created will be IVR-3, IVR-4 and so on. 

You will need to create another IVR for selecting the language options that you want to 

present to the callers. This will be the IVR that will greet all callers with Select 1 for 

English, 2 for Italian etc. In this example, we will have 2 language sets, Australian and 

Italian. Let us name this second IVR LanguageChoice. 

Being the second IVR you have created, this IVR will be known as IVR-3 in your 

extension_additional.conf file. Any subsequent IVR created will be IVR-4 and so on. 

In LanguageChoice, we will have 3 options. 

• Select 1 for Australian English 

• Select 2 for Italian 

• The third option is unannounced as it will be the time out option represented by 

the letter “t”. This option is the option the IVR will defaulted to if no choice is 

made or if the caller’s DTMF tone is not recognised by Trixbox. 

In choice 1, select the Custom App: radio button and enter the following in the box: 

Custom App 

custom-language_au,s,1 

In choice 2, select the Custom App: radio button and enter the following in the box: 

Custom App 

custom-language_it,s,1 

The third choice, instead of a number, you will enter the letter “t” in the option box, select 

Custom App: radio button and enter the following: 

Custom App 

custom-language_au,s,1 

Next, you will need to do a little editing of the extensions_custom.conf file. You need to 

add the following towards the end of your extensions_custom.conf. 

You will also need to create another IVR called MainMenu_IT with options to be 

presented in Italian. This is similar to your main IVR except that it is presented to the 

caller in Italian. This IVR being the third that you have created will be referred to as IVR-4 

in the extensions_additional.conf file. 

[custom-language_au] 

exten => s,1,Set(LANGUAGE()=au) 

exten => s,n,Playback(vm-dialout) 

exten => s,n,Goto(ivr-2,s,begin) 

[custom-language_it] 


exten => s,1,Set(LANGUAGE()=it) 



Save it and re-read the configuration change and you are done. 

Now you will need to change your inbound route to point to the LanguageChoice IVR. 

All calls will then be greeted by the LanguageChoice IVR and when the caller select 1 (or 

2), the IVR will call the appropriate exten. If the caller selected 1, the [customlanguage_au] 

will be selected and the language will be set to Australian English, an 

announcement in the appropriate language will be made and the caller will be sent to the 

MainMenu (or MainMenu_IT) for call options (you must also have recorded the call 

options announcement in the various languages). 

11.2.1 Complimentary multi-language MOH 

To take the multi-language IVR one step further, we can also have multi-language (or 

multi category) MOH to compliment the language. 

If the language used is Italian, it is safe to assume that the caller is Italian and would 

much prefer to listen to Italian music instead of Chinese or Arabic music. 

To do this you will need to have the appropriate MOH music in Trixbox. 

Assuming that you already have the default music category that comes with Asterisk, 

you will need to create a new music category using freePBX On Hold Music option. 

Let us call this new category ItalianTune. 

See the chapter relating to MOH in the later chapter of this document. 

Once you have created this category, upload a few MP3 Italian music/Tunes to it. If 

you want the tune to be randomised, click the Enable Random button and you are set 

to go. 

You will need to modify the 2 scripts above to the following: 

[custom-language_au] 

exten => s,1,Set(LANGUAGE()=au) 

exten => s,n, SetMusicOnHold(default) 



[custom-language_it] 

exten => s,1,Set(LANGUAGE()=it) 

exten => s,n, SetMusicOnHold(ItalianTune) 



You may have to restart asterisk after this (although it may not be necessary). 

Now you will have multi language IVR as well as the complimentary MOH. 


11.3 MULTI-TENANTS IVR 

(This is included in this section as an example only and is not part of this tutorial. However you 

can use this if you want to). 

While freePBX is a very versatile GUI front-end for Trixbox it does not have the facility to 

configure a PBX for multi-Tenants application. However, Trixbox (Asterisk) has the 

ability to do it with a little creative manipulation. 

In the chapter above we covered multi languages IVR handling by simply telling Trixbox 

what language to use. We are going to use the same principal to handle the Multi- 

Tenants requirement. 

In this example, I will use a Medical Centre in Malaysia, that I have the privilege of 

installing the Asterisk PABX for. There are 3 tenants involved – a General Practitioner, a 

Dental Practitioner and a Pathology representative. Each has separate incoming 

telephone numbers with their own receptionist but using a common PABX. 

I have picked this organisation as an example because it is small enough to write about. 

If I choose one of my bigger installations, this book will never see the light of day ☺ 

The main requirements are pretty straightforward: 

• Calls that come in through the different tenants should be directed to the 

appropriate IVR. One IVR for each tenant. 

• 12 extensions - 4 for Dental Practitioner, 3 for Radiologist and 5 for General 

Practitioners. 

• Voicemail for each receptionist. 

There are other requirements such as multi language prompts, after-hours ring groups, 

time conditions, Interlinking 2 Trixbox boxes between the Medical Centre and the 

Pathology Main Office in Kuala Lumpur etc., but I have only listed the major requirements. 

In this example I will only highlight the Multi-Tenants incoming route handling and 

restricting outgoing calls to the appropriate telephone lines only. 

The full case study can be found in the Appendix titled, Live Case Study at the end of 

this document. 

My first task was to install the base Trixbox as per the chapter on Installation. In this 

installation, there is no VOIP trunk involved. All telephone communications are through 

PSTN lines apart from the Trixbox inter-connect between the Medical Centre and the 

Pathology Main Office in Kuala Lumpur, which will be done via a DSL connection. 

PSTN Interface 

2 x TDM400 were installed in the PC designated for the PBX 

• 3 ports allocated to the General Practitioners (ports 1-3) 

• 3 ports allocated to Dental (ports 4-6) and 

• 2 ports allocated for Pathology (ports 7-8). 

IVR Prompts 

3 x IVR were created, one for each of the different tenant. 

• MedicalIVR (represented by ivr-3 in extensions_additional.conf) 

• DentalIVR (represented by ivr-4 in extensions_additional.conf) 

• PathoIVR (represented by ivr-5 in extensions_additional.conf) 


Each IVR will act as a traffic cop to direct calls to the appropriate extensions belonging to 

each tenant. 

Inbound Routings 

• ZAP channels 1,2 and 3 (3 PSTN numbers for GP) routed to MedicalIVR 

• ZAP channels 4,5 and 6 (3 PSTN numbers for Dentist) routed to DentalIVR 

• ZAP channel 7 and 8 (2 PSTN numbers for the Pathology representative) routed 

to PathoIVR 

Extensions 

1. Extensions 2000 to 2004 allocated to GP 

• 2000 for Receptionist/Clerk with voicemail 

• 2001 for Doctor A 

• 2002 for Doctor B 

• 2003 for Surgery Room 

• 2004 for Dispensary 

Change the context of these extensions, from [from-internal] to [from-internal-medical] 

2. Extensions 3000 to 3003 allocated to Dental 

• 3000 for receptionist/clerk with voicemail 

• 3001 for Dentist A 

• 3002 for Dentist B 

• 3003 Spare extension 

Change the context of these extensions, from [from-internal] to [from-internal-dental] 

3. Extensions 4000 to 4002 allocated to Pathology 

• 4000 for receptionist/clerk with voicemail 

• 4001 for Radiographer 

• 4002 for Assistant 

Change the context of these extensions, from [from-internal] to [from-internal-pathology] 

Outbound Calls Handling context 

Create 3 different call handling contexts in the extensions_custom.conf configuration 

file. 

• [from-Internal-medical] 

• [from-internal-dental] 

• [from-internal-pathology] 

The purpose of these contexts is to restrict extensions belonging to each tenant to be 

confined to telephone lines belonging to that particular tenant when making external calls. 

See How to restrict outgoing calls in the Calls Handling chapter earlier in this document. 

All of the basic set up requirements above are explained in the various chapters of this 

book and should not present any difficulty to complete. 

Once the basic is completed, the rest of the variations and extras can be handled as per 

the individual requirement. 


12 GET UNDER THE BONNET 

 

Config files manual tweaks in this guide are meant for advanced users. 

Unless you have a compelling reason to do so, users are advised against 

any manual config file tweaks. Lately, it seems that you must restart 

Asterisk everytime you make changes to the config file.) 

Just as you think that all is OK, you realized something else requires attention. This is 

true with Trixbox as well. 

To do this we may need to edit some configuration files (.conf) that reside both in the 

/etc/asterisk directory and /etc directories. Configuration files in the /etc/asterisk are 

generally editable through Config Edit. 

To start editing the .conf files we need to hover on the Asterisk tab and select the Config 

Edit selection from the dropdown menu as illustrated below. 

Select Config Edit (marked with arrow) and we will see a new screen with a list of all the 

.conf files, which can be edited manually. 

Scroll down the page to find the file that needs to be edited. 


12.1 EDITING THE .CONF FILES 

A number of .conf files may require editing to get asterisk to work, depending on the 

individual requirements. In this guide, as a base, we will be looking at sip.conf, iax.conf 

(just in case you want IAX, but not really necessary if you are just running SIP), 

indications.conf, enum.conf, extensions_custom.conf, features.conf, zaptel.conf 

and modprobe.conf (the last 2 files live in the /etc directory and may need to be edited if 

you have a FXO card e.g. X100P installed. We will look at them later on under 

Interfacing Asterisk to PSTN). 

12.1.1 sip.conf 

[general] 

port = 5060 ; Port to bind to (SIP is 5060) 

bindaddr = 0.0.0.0 ; Address to bind to (all addresses on 

machine) 

disallow=all 

allow=g729 ; I add this in sip_custom.conf instead 

allow=ulaw 

allow=alaw 

allow=gsm 

; the 2 lines below I added them in sip_custom.conf although 

; you can add them in sip.conf 

defaultexpirey=600 ; **Engin users: include this if necessary 

maxexpirey=3600 

; **Engin & BBP Global users: include this if necessary 

rtptimeout=60 

rtpholdtimeout=120 

qualify=yes 

progressinband=yes 

; If you need to answer unauthenticated calls, you should 

; change this next line to 'from-trunk', rather than 

; 'from-sip-external'. 

; You'll know this is happening if when you call in you get 

; a message saying "The number you have dialled is not in 

; service. Please check the number and try again." 

;context = from-sip-external ; Send unknown SIP callers to this context 

context = from-trunk ;

12.1.2 sip_custom.conf 

language=au 

videosupport=yes 

allow=g729 

allow=g723 

allow=h261 

allow=h263 

allow=h263p 

defaultexpirey=600 

maxexpirey=3600 

; **Engin users: include this if necessary 

; **Engin & BBP Global users: include this if necessary 

12.1.3 iax.conf 

In the general section, add the following lines if it is not already there: 

[general] 

delayreject=yes 

jitterbuffer=yes ; I placed this line in my iax_custom.conf instead 

mailboxdetail=yes ; I placed this line in my iax_custom.conf instead 

 

Every time you update freePBX, there is a good chance that changes or 

additions you have made in iax.conf will be overwritten. To avoid this, all 

additions I wish to make in iax.conf are made in iax_custom.conf 

12.1.4 iax_custom.conf 

; additions to iax.conf that would not get deleted if you 

upgrade freepbx 

;externip = or (one or the other NOT both) 

externhost = 

localnet = 192.168.1.0/255.255.255.0 

language=au 

delayreject = yes 

jitterbuffer = yes 

allow=g729 

allow=ilbc 

allow=g723 

12.1.5 Indications.conf 

In the general section ensure the following exist: 

[general] 

country = au ; (The default is country = us, therefore replace “us” with “au”) 

12.1.6 enum.conf 

[general] 

; 

; The search list for domains may be customized. 

; Domains are searched in the order they are listed here. 


; 

search => e164.arpa 

search => e164.org 

12.1.7 extensions_custom.conf 

If you have a ZAP trunk and you want to get an external line and be presented with a 

dial tone first before you dial the external line, you may add the following codes in the 

extensions_custom.conf file. 

The example below assumes that your ZAP is on channel 1. 

Immediately after the line following the [from-internal-custom] context, insert the 

following codes: 

exten => 0,1,Dial(ZAP/1/{EXTEN}) 

exten => 0,2,Hangup 

This will allow you to dial 0 and you will get a second dial tone from PSTN. Following 

which you can dial a PSTN number that you wish to call. 

 

. 

If you're using a PRI line, you need to enable 'overlapdial' in 

/etc/asterisk/zapata.conf for this to work. 

12.1.8 features.conf 

(Ignore this if you are using using Trixbox v2.6.x with the latest FreePBX) 

The default features.conf included with Trixbox is fairly minimal. It is recommended 

that you reflect the following in your features.conf file. The extra keywords will come 

in handy later on. 

; 

; Sample Parking configuration 

; 

[general] 

; do not manually enter parkinglot config information, use the parkinglot module 

; 

; the parking_additional.inc file is auto-generated by the Parkinglot Module, do 

; not hand edit that file 

#include parking_additional.inc 

#include features_general_custom.conf 

transferdigittimeout =>3 ; Number of seconds(in seconds) 

; to wait between digits when transferring a call 

courtesytone = beep 

; Sound file to play to the parked caller 

; when someone dials a parked call 

xfersound = beep 

; to indicate an attended transfer is complete 

xferfailsound = beeperr ; to indicate a failed transfer 

;adsipark = yes 

; if you want ADSI parking announcements 

;pickupexten = *8 ; Configure the pickup extension. Default is *8 

featuredigittimeout = 800 ; Max time (ms) between digits for 

; feature activation. Default is 500 

[applicationmap] 

#include features_applicationmap_additional.conf 


; *** IMPORTANT NOTE *** 

; The original blindxfer was '#', and has been changed to '##' to avoid 

; issues with sending DTMF '#' to remote parties. 

[featuremap] 

blindxfer => ## 

disconnect => ** 

automon => *1 

;atxfer => *2 

; Blind Transfer (default is # - change if having 

; problem with external DTMF that requires #) 

; Disconnect Call 

; One Touch Record 

; Attended Xfer 


13 CHECK YOUR SYSTEM 

After doing all of the above, you will no doubt feel frustrated if your VOIP does not work. 

The following are some basic steps for you to do – at least to ensure that your installation 

is running correctly and is registered to the appropriate VSP before you start doing 

anything more adventurous. 

There are two places where you can obtain a quick health report of your system – the 

System Process Status and the Asterisk Info screens. 

 

If you don’t have the Asterisk Info option, refer to the NOTES section on how to 

obtain this module. 

13.1 SYSTEM PROCESS STATUS 

First you must ensure that asterisk is well and running or you may not be able to make or 

receive any call. 

If all processes are running then your battle is half won. 

In Trixbox 2.6, the information can be obtained straight from the Admin Maintainance 

screen as soon as you change user mode to Admin mode – see red arrow in the 

illustration below.. 

If any of the four services aren’t running, chances are something is wrong and your 

Asterisk will not be working correctly or not working at all. A probable cause can be 

wrong configuration of your zaptel drivers. If you don’t have a zaptel devise, you should 

try to disable the related .conf files. 

Ensure that the minimum requirements have been met. If the minimum requirements and 

the right type of hardware requirements have been met, and yet it is still not working, then 

maybe its time to start the installation again. 

Next, select and click Asterisk Info to verify your status. 


13.2 ASTERISK INFO 

From freePBX, select the Asterisk Tab and click on the Asterisk Information option as 

shown by the red arrow. 

When presented with the info page, select Show All. See illustration below. 

You will then be presented with the following screen. 


Scroll down to the SIP (or IAX if you are using IAX) section and ensure that you are 

registered to your trunks. You will only be able to make and receive external calls – 

hopefully - if you are registered. 

If all trunks are registered, you can be almost certain that at the very least, your Trixbox 

installation is working (although not necessarily that you can make or receive calls). 

In the majority of cases, if you have done everything as described in the preceding 

chapters, the chances of your installation working are pretty good. However there are 

always times when you think that the weather is going to be fine and yet you are pelted by 

a thunderstorm. 

If you are not able to make or receive calls, the most common causes are Trunk 

Registrations, the choice of Codec, Routings, and Dialling rules errors. 

Routings and Dialling rules are something that needs to be thought out logically with no 

simple way of determining, as different people has different requirements and different 

VSP have different dialling rules. 


14 INTERFACING ASTERISK TO PSTN 

A new issue has raisen its ugly head with Trixbox 2.2.4. Building Trixbox 2.2.4 from the 

ISO broke the zaptel install module. As a result, zapel module will not be found and 

consequently the zaptel cards will not be detected 

To remedy this, log in as root and do the following at the command line: 

yum clean all 

yum install zaptel* 

14.1 A100P FXO PCI CARD 

The X100P card is no longer being made, but there are many X100P compatible in the 

market – some good and some not so good. The cards I am using is the A100P variety 

from Openvox. The 3 A100Ps I use for testing and POC (in different boxes of course) 

have been working flawlessly for the last 7 months; unfortunately, since Asterisk no 

longer will be supporting the X100P officially, Openvox has stopped its production. 

. 

This card allows you to connect a POTS (plain Old Telephone System) line to your 

Asterisk box. 

OpenVox A100P Card 

If this card is added after Asterisk has been configured, it may be necessary to configure 

it by using the zaptel card auto-config utility so the correct zaptel driver will be set up. To 

do that, enter the following from the command line. 

Setup-pstn 

Restart Asterisk 


After restart run the following: 

genzaptelconf –s –d –v –c au 

Restart again. 



14.2 CREATE A ZAP TRUNK 

Next go into the freePBX web interface to create a trunk and you will notice that there is 

already a trunk called ZAP/g0. You need to edit this. 

1. Enter the phone number for you pots line in the Caller ID field 

2. Enter 1 for Maximum channels 

3. Set a dial rule you want for this trunk 

4. Select an outbound dial prefix to select this trunk when dialling 

5. Set the Zap Identifier to 1 (the default is g0) 

Once the card is configured, you must add a route for Incoming Calls or asterisk will not 

answer this line 

Click on Incoming Calls in freePBX and set up an incoming route. To make outbound 

calls you will need to set an outbound route as well. 

If you have this card installed, you may need to edit the following files (usually you don’t 

have to); zapata.conf, zapata-auto.conf, zaptel.conf and modprobe.conf. The last 2 

files live in the /etc directory – use a text editor (I use nano) to edit them as you cannot 

edit them through Config Edit. 

14.2.1 zapata.conf 

Under [channels] add the following lines: 

[channels] 

busydetect=yes 

busycount=6 

The above is to ensure that the line is hung up if there is no answer or busy after 6 

busy tones. 

For my installation to function correctly, I have also changed the following setting to 

obtain a good compromise on volume/echoing: 

rxgain=4 (you may have to experiment a little with this setting) 

txgain=2 (you may have to experiment a little with this setting) 

14.2.2 zapata-auto.conf 

Ensure the following exist in Zapata-auto.conf. It is located at the end of the file. 

context=from-zaptel 

group=0 

channel => 1 

Leave the rest of the file as it is. 

14.2.3 zaptel.conf 

Change the loadzone and defaultzone to “au” 

# Global data 

loadzone = au 

defaultzone = au 


14.3 OPENVOX A400P FXO/FXS CARD 

Like the A100P, this card allows you to connect a POTS (plain Old Telephone System) 

line to your Trixbox. Unlike the A100P, this card has 4 module ports that can be loaded 

with FXS or FXO modules. Channel 1 is the top RJ-45 on the back of the card. The 

illustration below is the OpenVox A400P card and 2 fxo modules. This card is 100% 

compatible with the TDM400 and is very reliable. 

If this card is installed after Asterisk has been loaded, you will need to configure it just like 

the A100P by using the following command on the command line: 

setup-pstn 

Restart Asterisk 


After restart run the following; 

genzaptelconf –v –c au (au is for Australia) 

Restart again. 

shutdown –r now 


Next, using config edit, look in the zapata-auto.conf file and you will see a list of all 

your channels in your A400P. Set up the trunks as trunks and the extensions as 

extensions in freePBX. 

When you open the zapata-auto.conf file, it will look something like the illustration 

below. 


; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit 

; Zaptel Channels Configurations (zapata.conf) 

; This is not intended to be a complete zapata.conf. Rather, it is intended 

; to be #include-d by /etc/zapata.conf that will include the global settings 

; 

callerid=asreceived 

; Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" 

signalling=fxs_ks 

; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1 

context=from-pstn 

group=0 

channel => 1 

; channel 2, WCTDM, inactive. 



If in the illustration it shows channel 1 is your Zap extension then add a zap extension 

for channel 1 in freePBX and if it shows your Zap trunk is channel 2 you should 

create a zap trunk for channel 2 in freePBX. 

Once this is done, reboot your PC and when Asterisk starts, use freePBX to add a 

route for incoming calls or asterisk will not answer your trunk. Similarly, to make 

outbound calls you will need an outbound route. Set them up as per setting up routes 

in the earlier chapters of this document. 

If you have this card installed, you may need to edit the following files: zapata.conf 

and zaptel.conf as per the X100P card in the previous section. 

14.3.2 modprobe.conf 

You may need to edit the modprobe.conf to add the necessary option for usage in 

Australia. 

You may use nano or WinSCP3 to edit /etc/modprobe.conf. Simply add the line in 

red at the bottom of the file as illustrated below. 

alias eth0 natsemi 

alias snd-card-0 snd-intel8x0 

options snd-card-0 index=0 

install snd-intel8x0 /sbin/modprobe --ignore-install snd-intel8x0 && /usr/sbin/alsactl restore >/dev/null 2>&1 || : 

remove snd-intel8x0 { /usr/sbin/alsactl store >/dev/null 2>&1 || : ; }; /sbin/modprobe -r --ignore-remove snd-intel8x0 

alias usb-controller uhci-hcd 

options wctdm opermode=AUSTRALIA 

Once this is done, execute the following command. 

modprobe wctdm 

If there is no error, 

reboot 

To ensure that you have the right mode operating, do the following at the command 

line: 

grep "AUTO FXO" /var/log/messages 

or 

dmesg 


You should see something like this: 

Mar 10 13:30:09 asterisk1 kernel: Module 0: Installed -- AUTO FXO (AUSTRALIA mode) 




If you see an FCC mode then you will have an impedance mismatch. 

 

As of Zaptel Drivers 1.2.4, by selecting opermode=AUSTRALIA the zaptel drivers 

automatically add the "boostringer=1, fxshonormode=1". Also see Appendix E.3 

(Users Suggestions) 

14.4 IS THE A100P/A400P INSTALLED 

At the command line, run lspci and amongst others, you should see the following; 

00:0f.0 Communication controller: Motorola Wildcard X100P 

Determine if there is any conflict with other PCI hardware. If there is, you probably will 

need to replace or use another motherboard. If there is no conflict, make sure you have 

the kernel sources installed and rebuild the zaptel component. 

setup-pstn 

genzaptelconf –v –c au (au for Australia) 

ztcfg 

Once done, run zttool on the command line to ensure the the card is functioning. 

To confirm that zaptel is actually running, after reboot, log in to Asterisk and at the 

command prompt issue the following command to configure zaptel. 

genzaptelconf 

Then issue the following command; 

modprobe wcfxo 

If there is no error then your card has been detected. Prove it by issue the following 

command: 

dmesg 

You should see the following in amongst the messages; 

wcfxo: DAA mode is 'FCC' 

Found a Wildcard FXO: Wildcard X100P 

After that restart amportal. 

Amportal restart. 

Using your browser on another PC, connect to TRIXBOX and using the Asterisk Info 

option, check to see if zaptel is loaded. 


Chances are, Zaptel will be found. 

14.5 CALLER ID (CID) USING ZAP DEVICE 

The illusive CID has been a talking point and hair pulling exercise for many users. It has 

always been the case of “I have my ZAP trunk working fine but it is not sending the CID to 

my Softphone or SIP Phone”. 

The simple answers to this: 

• In Australia, for you to get Caller ID is actually having to apply to your Telco to 

have it activated on your line – this is not activated by default by your Telco 

(Optus or Telstra). If caller ID is not activated on your line, you will not get CID. 

• If you already have caller ID activated and still don’t get caller ID, it’s time to have 

a look at the ZAP configuration file. Typically you may have to look at 

zapata.conf and/or zapata-auto.conf. 

Naturally if you are using an X100P clone for your FXO, the following exercise will 

probably be futile as the X100P, even at the best of time will probably not handle the CID 

properly. However if you are using a more substantial device such as the A400P or 

TDM400, then this may possibly help you towards getting the CID passed correctly. 

(caveat: This solution may not necessarily work for your installation but you have nothing to 

loose if you try) 

A participant (Shane B) in the Whirlpool Forum provided the solution below that will 

correctly pass the CID to Asterisk and your telephones: 

You may need to set the following switches in your zapata.conf and zapata-auto.conf. 

14.5.1 zapata.conf 

usecallingpres=yes 

callwaitingcallerid=yes 

threewaycalling=yes 

usecallerid=yes 

hidecallerid=no 

relaxdtmf=yes 


The following switches may need to be added to the existing ones. 

useincommingcalleridonzaptransfer=yes 

adsi=yes 

sendcalleridafter=2 ;you may need to add this switch 

After the above are done, restart Trixbox: 

Amportal restart 

If you are lucky, it should work. If not, you haven’t really lost anything. 


14.6 SIPURA SPA3000/3102 AS A PSTN INTERFACE 

To those new to the SPA3000, there is a simplified installation and configuration 

instruction by JMG Technology. While it is directed mainly at standalone ATA users, it 

gives a good insight of the Sipura SPA3000’s capabilities. 

I have come across a few people in the various forums wanting to use their Sipura SPA- 

3000s as FXO front-end to their TRIXBOX boxes. To help them in their endeavours, I've 

put the following together, as no one single source of information that I've found so far 

has a config that would actually work for me. 

14.6.1 Log in to SPA3000/3102 

Login to your SPA-3000/3102 as admin/advanced. 

Before you change anything, I'd suggest taking a snapshot (i.e. just save the .html 

page) of your current SPA-3000 configuration, just in case you ever need to refer 

back to your own customisations. 

If you're not already running the latest SPA-3000 firmware, then upgrade it to the 

latest version (at the time of writing, it's 3.1.5a). Take another snapshot for good 

measure. Nothing should have changed in your settings, except that you have a few 

extra options that you didn't have before. 

Now reset SPA-3000 back to factory defaults, because I'm only going to list the 

minimum changes required to keep things simple. Take another snapshot now too, in 

case you ever want to know what the defaults were. 

14.6.2 Change the settings 

System tab 

DHCP: No 

Static IP: something on your local subnet e.g.; 192.168.1.200 

NetMask: 255.255.255.0 

Gateway: your router's IP address e.g.; 192.168.1.254 

Primary DNS: your ISP's primary DNS address e.g.; 203.12.160.35 

Secondary DNS: your ISP's secondary DNS address e.g.; 203.12.160.36 

Regional tab 

Dial Tone: 400@-19,425@-19;10(*/0/1+2) 

Busy Tone: 425@-10;10(.4/.4/1) 

Reorder Tone: 425@-10;10(.2/.2/1) 

Ring Back Tone: 400@-19,425@-19,450@-19;*(.4/.2/1+2+3,.4/.2/1+2+3,0/2/0) 

Ring 1 Cadence: 60(1.5/3.4) 

Ring 3 Cadence: 

60(1.5/3.4,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/. 

2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2,.4/.2,.4/2) 

CWT8 Cadence: 30(.2/.2,.2/4.4) 

Hook Flash Timer Min: .07 

Hook Flash Timer Max: .13 

FXS Port Impedance: 220+820||120nF 

 

Delete all the Vertical Service Activation Codes. 

Line 1 tab 

Proxy: IP address of your Asterisk box e.g.; 192.168.1.100 

Register Expires: 60 

Display Name: Whatever 


User ID: Asterisk extension number e.g.; 2000 

Password: password for that extension 

Silence Threshold: medium 

DTMF Tx Method: INFO 

Hook Flash Tx Method: INFO 

Dial Plan: (**|*xx|**x.|000|0011xxxxxxxx.|0[23478]xxxxxxxx|09xxxxxx|1100 

|122[135]|1222xxxxxxx|12510[12]|12554|1[38]00xxxxxx|13[1-9]xxx 

|2xx|393xxxxxx|3xxxx.|[4689]xxxxxxx|899060xxxxx.) for example 

(**|*x.|x.|**x.) will work, but I like to do a bit of sanity checking, etc. 

PSTN Line tab (method 1) 


Register: no 

Make Call Without Reg: yes 

Ans Call Without Reg: yes 

Display Name: No name 

User ID: PSTN 

Password: password 

Silence Supp Enable: no 

Echo Canc Enable: no 

Echo Canc Adapt Enable: no 

Echo Supp Enable: no 

FAX CED Detect Enable: yes 

FAX CNG Detect Enable: yes 

FAX Passthru Codec: G711u 

FAX Codec Symmetric: no 

FAX Passthru Method: None 


FAX Process NSE: no 

Dial Plan 1: (S0) for example - it could be your phone number 

VoIP Caller Default DP: none 

PSTN Ring Thru Line 1: no 

PSTN CID For VoIP CID: yes 

PSTN Answer Delay: 2 

PSTN Ring Thru Delay: 3 

PSTN Ring Timeout: 4 

PSTN Hook Flash Len: .1 

Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2) 

FXO Port Impedance: 220+820||120nF 

On-Hook Speed: 26ms (Australia) 

Using method 1 above, it will be necessary for you to create an inbound route using 

the 0298765432 (or whatever you have defined) as your inbound route DID number. 

As this incoming call is identified by the (S0) in the dial plan 1 setting 

of your SPA3000, this can be directed to any extension by the inbound route. 

PSTN Line tab (method 2) 


Register: no 

Make Call Without Reg: yes 

Ans Call Without Reg: yes 

Display Name: No name 

User ID: PSTN 

Password: leave blank 

Silence Supp Enable: no 

Echo Canc Enable: no 

Echo Canc Adapt Enable: no 

Echo Supp Enable: no 

FAX CED Detect Enable: yes 

FAX CNG Detect Enable: yes 

FAX Passthru Codec: G711u 

FAX Codec Symmetric: no 

FAX Passthru Method: None 



FAX Process NSE: no 

Dial Plan 1:(S0) e.g. (S0

nat=no 

port=5061 for example 

type=user 

username=PSTN 

Leave "Register String" empty 

14.6.4 SPA3000 as an outbound PSTN Trunk 

To make PSTN calls using the SPA3000 you simply select the SPA3000 trunk, which 

you have named Telstra (see trunk name that you have given to this trunk above) as the 

first choice in your outbound route. 

Also see Eliminating echo problems in Appendix E.3 in Sipura SPA-3000 

14.7 SPA3000 REMOTE MANAGEMENT 

The SPA3000 can be managed remotely. To enabled remote management the 

following must be set as per the illustration below: 

Naturally you will have to use the appropriate IP addresses, port and your own 

password etc. The illustration above is for my system. 

 

If you are managing your SPA3000 remotely, you may encounter some 

difficulties displaying the pages or sometimes it just seems to hang. To solve this 

problem, you can try changing your TCP setting (I use TCP Optimizer) by 

setting the RWIN to something smaller eg. 10xMTU, i.e if your MTU is 1500, try 

setting your RWIN to 15000. 


14.8 USE STUN SERVER ON SPA3000 

To use a STUN server with the SPA3000, simply select the SIP tab on Admin 

Admanced mode and make the following changes. I use stun.xten.com, you may use 

any STUN server of your choice. 

a. Select SIP Tab 

b. Handle VIA received: no 

c. Handle VIA rport: no 

d. Insert VIA received: no 

e. Insert VIA rport: no 

f. Substitute VIA Addr: yes 

g. Send Resp To Src Port: yes 

h. STUN Enable: yes 

i. STUN Test Enable: no 

j. STUN Server: stun.xten.com (I use stun.xten.com, you may use 

whichever stun server you like) 

k. NAT Keep Alive: 15 


15 PSTN TO VOIP GATEWAY 

 

If you are using the later versions of Trixbox, the following instruction may not 

apply as you can create DISA from FreePBX option without having to resort to 

the manual configuration below – see DISA, 

Now that the SPA-3000 has been set-up to accept PSTN, TRIXBOX can be turned into a 

PSTN to VOIP Gateway. You will not need a SPA-3000 if you have an FXO card 

installed. 

I need such a gateway because I often have to call overseas from my office. Needless to 

say, we don’t make private overseas calls from the office phone. I overcame this by 

calling my Asterisk box from the office and when it answered, I can then use asterisk to 

call overseas. 

I accomplished this through DISA – I have resorted to this manual method because in the 

initial release of Trixbox , DISA did not work, as I would like it to. This has since been 

rectified 

To do this, I manually edit the extensions_custom.conf, by adding the following entries 

at the end of the file, 

[custom-pstn2voip] 

; I use this to call VOIP from PSTN 

exten => s,1,Answer 

exten => s,2,DigitTimeout(5) 

exten => s,3,ResponseTimeout(10) 

exten => s,4,Authenticate(my password) 

exten => s,5,DISA(no-password|from-internal) 

I have Digital Receptionist answers all incoming calls (refer to Digital Receptionist in the 

previous section). 

In the previous section I have 5 options in Digital receptionist e.g. Press 1 for Ben, 2 for 

Rohani, and 3 for Norsurya. I did not announce the 4th option as I use it for DISA and is 

only known to me, besides, it requires a password. 

For option 4, I chose the radio button next to Custom App. In the custom app field I 

entered custom-pstn2voip,s,1 

Custom App 

custom-pstn2voip,s,1 

When 4 is pressed, followed by the # key, during the announcement, the digital 

receptionist will ask for a password. On entering the correct password, the one defined 

in the example above, a dial tone will be heard. At this stage, you can start dialling the 

number you wish to call. 

Option 5, (which I have changes to the letter ‘t’, for timeout) is invoked if the caller does 

not select any option or if the DTMF tone is not recognised. This option can be directed 

to a Ring Group or to a human receptionist if you are using Asterisk in a business 

environment. 


16 DISA 

16.1 DISA - HAND CODING METHOD 

I use this function in my PSTN to VOIP gateway above. 

DISA (Direct Inward System Access) allows you to dial in from outside to the Asterisk 

switch (PBX) to obtain an "internal" system dial tone. You can place calls from it as if 

they were placed from within. 

Normally you can do this via the Digital Receptionist and enter the unannounced option 

number known only to selected callers. 

The caller enters his/her password, followed by the pound sign (#). If the password is 

correct, the user is then given system dial tone on which an external call may be placed. 

This type of access has SERIOUS security implications, and GREAT care must be taken 

NOT to compromise your security. 

The example below is how I have implemented my DISA. 

In my "MyMenu" that says, “Hi, you have reached the Sharif’s household. Press 1 for 

Ben, 2 for Rohani, 3 for Norsurya etc…” I have created an extra option – option 4 (best if 

you use a few digits) that has an "Action" of "Custom App", which reads, "custompstn2voip,s,1". 

Then in extensions_custom.conf, I have added (do this towards the end of the file). 

[custom-pstn2voip] < -- you can name it to whatever you like 


exten => s,2,DigitTimeout(5) 

exten => s,3,ResponseTimeout(10) 

exten => s,4,Authenticate(12345678) 

exten => s,5,DISA(no-password|from-internal) 

Change "12345678" to whatever digits you want your "DISA password" to be. 

The above is one way of doing it (the hard way). The easier way of doing it is through 

freePBX - see below. 


16.2 DISA – USING FREEPBX 

Those using Trixbox with freePBX as the User Interface will be able to configure DISA 

from freePBX without having to do the above manual configuration. The same thing can 

be achieved through freePBX as illustrated in the screenshot below. 

I have populated the various fields thus: 

• DISA name: PSTNVoIP (This is the name I gave to this application) 

• PIN: 1234 (or whatever you think is appropriate. This pin number will be asked 

when you want to use this service) 

• Response Timeout: 10 (time needed for a response from the you in seconds) 

• Digit Timeout: 5 

The rest you can leave at the default value. 

When you set up your digital receptionist, in the DISA dropdown selection, simply select 

PSTNtoVOIP as your DISA application. 

If you are using this method, there is a slight catch. When you choose the DISA option to 

call a number, you will be greeted with “Please enter your password followed by the 

pound key” and immediately you will get a dialtone. You may start dialling the password 

followed by # followed by the telephone number. 

E.g. If you have chosen to have a password e.g. 12345, and the number you wish to dial 

is 0292345678, you must dial the number thus: 12345#0292345678 without waiting for 

another dialtone, otherwise it will not dial out correctly and you will be wondering what 

happened. I hope this will be fixed up soon. 


17 OPERATOR PANEL 

Operator Panel is a switchboard type application for the Asterisk PBX. Typically, it runs 

on a web browser. It is able to display information about your PBX activity in real time. 

The Flash Operator Panel is the one that comes with TRIXBOX. 

Apart from Flash Operator Panel, There are a number of operator panels available. One 

that comes to mind can be found at SOFTPEDIA and another is HUDLite from Fonality. 

HUD will integrate nicely with TRIXBOX and the support for HUD is already integrated 

within TRIXBOX. 

17.1 FLASH OPERATOR PANEL (FOP) 

This is the standard Operator Panel that is already part of TRIXBOX – some loved it 

while some hated it. 

The layout is configurable (button sizes and colours, icons, etc). You can have more than 

100 buttons active per screen. It also supports contexts: you can have one server 

running and many different client displays (for hosted PBX, different departments, etc). 

It can integrate with CRM software, by popping up a web page (and passing the CLID) 

when a specified button is ringing. 

FOP can be activated either from within freePBX by clicking on the Panel tab on the top 

of the freePBX screen or from TRIXBOX User Mode Welcome screen. 

The following information are displayed on FOP: 

• Which extensions are busy, ringing or available 

• Who is talking and to whom (CLID, context, priority) 

• SIP and IAX registration status and reachability 

• Meetme room status (number of participants) 

• Queue status (number of users waiting) 

• Message Waiting Indicator and count 


• Parked channels 

• Logged in Agents 

Functions you can perform on FOP: 

• Hang-up a channel 

• Using drag-&-drop to transfer a call 

• Initiate calls by drag-&-drop 

• Barge in on a call using drag-&-drop 

• Set the caller id when transferring or originating a call 

• Automatically pop up web page with customer details 

• Click-to-Dial from a web page 

• Mute/Unmute meet-me participants 

For more details refer to: http://www.asternic.org/ 

17.1.1 Setting the Admin Password 

It is recommended to change the FOP password to something easy and simple to 

remember such as the number 2 for example. The simple method is by logging in to 

your asterisk box either remotely using putty or directly on your box console. 

In this example, Putty is used to log in remotely to TRIXBOX . Once logged in, 

change the directory to /var/www/html/panel 

cd /var/www/html/panel 

Using nano as the editor, open the configuration file op_server.cfg 

nano op_server.cfg 

Go to the line that says security code=passw0rd 

(in FOP that comes with TRIXBOX , the default password is “passw0rd”) 

Replace the “passw0rd” with the password of your choice. 

security_code=cowscomehome 

Close off nano and putty. Open your web browser and go to FOP. You should now 

be able to click on the little lock, put in your password and you will see it lock up. Now 

you can begin to have some fun. 

17.1.2 Hang-up a Call 

You can end a call by clicking the big red button. 

17.1.3 Transfer a Call 

Simply drag the little telephone over to the new extension and it will transfer the call. 

17.1.4 Initiate a Call 

If you are lazy, like me, not bothered to pick up the phone and dial, you can drag and 

drop your little telephone to another extension it will start by ringing your phone, when 

you pick up and then it will ring the other extension. Once the destination phone is 

picked up it will initiate the call. 

17.1.5 Barge in or Create a Conference 

Say there is a call already going and you want to jump in. Well all you have to do is 

grab your little phone in the flash panel and drop it into the conversation and it will 

connect you. Great if you want to see if your employees are making lots of private 

calls. 


17.2 HUDLITE 

HUDLite from Fonality integrates very nicely with Trixbox and the HUD manager is 

already incorporated in Trixbox. It has most of the necessary functionalities (if not all) of a 

modern operator switch panel where you can handle such tasks as: 

• Answer Calls 

• Transfer calls 

• Hang up calls 

• Put calls on hold etc. 

• Monitor status of extensions just like the default FOP that comes with TRIXBOX 

and more. 

To install HUDLite server, you must log in to Package (Trixbox Packages Tab) and select 

to install HUDLite server from the Package Manager. 

Once that is installed, you may proceed to set up HUDLite Admin. 

17.2.1 Setting up HUDLite Admin 

To set up HUDLite Admin, you will need to mouse ove the Asterisk Tab and select 

HUDlite Admin selection from the dropdown menu as shown below. 

When clicked with the mouse, you will be presented with the following screen where 

you may populate it with new devices.. 

Click on HUDlite Admin to populate it. 

You will be presented with the screen as illustrated below. 


To populate the HUDlite Admin, click on New device and the manage device fields 

will be expanded. This is where you enter the credentials of the extensions that you 

want to register with HUD Manager. 

In the manage device fields I have entered the following: 

Device: SIP2001 (the device for my extension) 

Ext: 2001 (my extension) 

Name: Ben Sharif (that’s my name if you don’t know) 

Username: 2001 (that’s the extension ID) 

Password: 2001 (I like it real simple) 

The other three fields are not mandatory and you can leave them blank. 

Click save. 

If you have more extensions to register, click on New device again and repeat this 

process as many extensions as you have. 

Once this is done, you may proceed to install HUDLite. 

17.2.2 Setting Up HUDLite 

After downloading, simply click on the EXE file and HudLite will be installed in your 

Windows XP machine. Select the Typical setup option. 


Once installed, run HUDLite you will be prompted to enter your Username and 

Password: 

Enter your username and password as required. I would enter one of my extension 

number here and the extension password. 

In my case I use my main office extension 2000. 

Username: 2000 

Password: 2000 

Click Next 

Next you will be asked for the server details. This is your TRIXBOX credentials. 

However in the password field, don’t use your server password, instead use HUDLite 

default password of password. 

. 

In the Server name I entered my Trixbox IP address: 192.168.1.100 

Server password, I simply use the default password: password. 

(I strongly urge that you use the default password for this initial installation. You can change 

it later when you are more comfortable with tinkering around) 

Then click Finish. 

HUDLite will now start. Initially you will see just a blank hudlite screen like the 

illustration below, as it will take a few seconds for it to gather some details. 


After a few seconds, it will populate all the extensions that you have added in the 

HUD Manager. 


If any of the extension is being called or making a call, you will see it highlighted just 

like the main office extension on the top left corner (the main number rang as I was 

capturing the illustration). 

17.2.3 What if it does not work 

If after following the above instruction, and it does not work, make sure that you have 

the correct values in the setup above. 

Click on File -> Setting on the HUDLite screen, expand the Advanced setting and 

make sure that the fields are correctly set as the set-up instructions above. 

Username: 2000 (you may have different extension) 

Password: 2000 (you may have different password) 

Server Name: 192.168.1.100 (your Trixbox IP may be different from mine) 

Server Password: password (it must be password) 

Leave the other 3 fields at the default value. 

 

The default password for HUD admin is password. 


18 CALL PARKING AND TRANSFER 

Trixbox provides this facility through the Parking Lot facility of freePBX. If you want to 

allocate other extensions rather than the one pre-configured, simply change the 

appropriate Parking Lot extension number in the configuration screen. For this purpose 

we will leave it at the default extension number 70. 

18.1 CALL TRANSFER - MANAGED 

The definition of this facility is found in parking_additional.inc and it contains the 

following lines: 

parking_additional.inc 

;*** WARNING: DO NOT HAND EDIT THIS FILE IT IS AUTO-GENERATD *** 

; 

parkext => 70 

; What ext. to dial to park 

parkpos => 71-79 

; What extensions to park calls on 

context => parkedcalls ; Which context parked calls are in 

parkingtime => 60 

; Number of seconds a call can be 

; parked for (default is 45 seconds) 

The above script is being called from feature.conf. 

; 

; Sample Parking configuration 

; 


[general] 

; do not manually enter parkinglot config information, use the parkinglot module 

; 

; the parking_additional.inc file is auto-generated by the Parkinglot Module, do 

; not hand edit that file 

#include parking_additional.inc 

#include features_general_custom.conf 

[applicationmap] 

#include features_applicationmap_additional.conf 

; *** IMPORTANT NOTE *** 

; The original blindxfer was '#', and has been changed to '##' to avoid 

; issues with sending DTMF '#' to remote parties. 

[featuremap] 

blindxfer => ## 

disconnect => ** 

automon => *1 

;atxfer => *2 

; Blind Transfer 

; Disconnect Call 

; One Touch Record 

; Attended Xfer 

The following are other options that you can include in the general section of 

features.conf if necessary. 

transferdigittimeout =>3 

courtesytone = beep 

; Number of seconds to wait between digits 

; when transferring a call (in seconds) 

; Sound file to play to the parked caller 

; when someone dials a parked call 

xfersound = beep 

; to indicate an attended transfer is complete 

xferfailsound = beeperr ; to indicate a failed transfer 

;adsipark = yes 

; if you want ADSI parking announcements 

;pickupexten = *8 ; Configure the pickup extension. Default is *8 

featuredigittimeout = 800 ; Max time (ms) between digits for 

; feature activation. Default is 500 

Each call parked will be parked at that extension +1 e.g. if you have 3 calls that you need 

to park, the first call parked will be parked at ext. 71, the next call will be automatically 

allocated ext. 72 and the next ext. 73 and so on. 

This is handy if you want to manage the calls to be transferred. After parking the call, you 

may call the person you are transferring it to and tell him/her that there is a call on 

extension 71. The person can then dial 71 and get the parked call. 

18.1.1 How is it done 

When you answer a call and want to transfer that call, do the following: 

1. Dial ##70 - the call will be transferred to a park extension and the number 

will be announced to you. In this case it will be 71. After the announcement, 

you may hang up and dial the extension of the person you wish to transfer 

the call to and inform him to dial 71 to accept the call. 

2. While all this is happening, the caller will be listening to the music on hold. 

3. If for any reason the person refused to take the call, you then hang up and 

dial 71 to pick up the call back to you and speak to the caller making the 

usual excuses “I am afraid Mrs. Roberts, Dr. Roberts is busy with his nurse and 


his office door is being locked from the inside” ☺ and the next time Dr. Roberts 

will always accept the calls from his wife. 

18.2 CALL TRANSFER – BLIND 

To do a blind call transfer (where you simply transfer the call and bad luck if the extension 

does not answer), you simply dial ## followed by the extension that the call is meant for. 

E.g. to transfer to extension 2001, you simply dial ##2001 and hang up. 

18.3 PUT A CALL ON HOLD 

Just like the managed call transfer above, you dial ##70 and after the parked extension 

announcement, you hang up. This is simply a work around for those that do not have the 

hold function on the telephone. 

When you are ready to take the call, dial 71 (or whatever the park extension is) and the 

call will be picked up again. While the call is being parked, the caller will be listening to 

your MOH music. 

18.4 CALL PICKUP 

Call Pickup is the ability to pick up a call on a phone that is ringing from other than the 

ringing phone. E.g. you are sitting at your desk and the phone in the next office is ringing 

but there is no one there to answer the phone. You then use the phone on your desk to 

pick up the call. 

There are two type of call pickups:- 

• Group call pickup 

This allows you to pickup a call from any ringing phone that is in the same pickup 

group as you. However, if there were more than one phone ringing, you would 

have no control over which call will be transferred to you. 

• Directed pickup 

This allows you to pickup a call that was made to a specific extension, such as 

when you are in another office and a phone rings in the general direction of your 

office. You are wondering if it is your phone that is ringing. In this case you 

simply dial the pickup number and your extension number. If the ringing phone is 

your phone, the call will be transferred to you. 

Group call pickup is typically invoked by dialling ** from another phone in the call pickup 

group. 

You can also change to *8 or to something else if you wish. 

see http://www.voip-info.org/wiki/view/Asterisk+config+features.conf 

18.5 FILTER YOUR INCOMING CALLS - ONLY ACCEPT KNOWN 

CALLS 

(Good for filtering Telemarketers and ex-girlfriends) 

For this purpose, the Privacy Manager option of the extension can be invoked, however I 

found that caller often get confused with the need to enter the 10 digits phone number as 

often they may have only 9 digits or more than 10 digits. 

I have opted for the Nerd Vittles method. 


There is a good article on this site http://mundy.org/blog/index.phpp=66 relating to filtering 

incoming calls and also some discussions here http://forums.whirlpool.net.au/forumreplies.cfmt=418909 

This feature will force callers to record their names before the calls will be parked. The 

extension that you nominate will ring and on answering, you will be informed that there is 

a call waiting for you in a parked extension e.g. 71. 

After receiving the call and if you wish to take the call, hang up and dial 71 and you will be 

connected to the caller. Otherwise, just hang up and the caller will just have to leave a 

message in the voicemail. Real nifty ☺ 

This is how I did it with my Asterisk. Since I am pretty simple myself, I just take the low 

road make it really simple. 

I add the following scripts in my extensions_custom.conf 

[from-internal-custom] 

; I create this extension for silent Auto attendant 30-10-2005 

exten => 111,1,Answer ; 

exten => 111,2,Goto(who-r-u,s,1) 

(Note: DO NOT create extension 111 in freePBX. This is only a 

dummy extension) 

At the end of the extensions_custom.conf, I added the following 2 blocks of extra 

codes. 

; experimental as per above 30-10-2005 

[who-r-u] 

exten => s,1,Background(privacy-unident) 

exten => s,2,Background(vm-rec-name) 

exten => s,3,Wait(2) 

exten => s,4,Record(/tmp/asterisk-stranger:gsm|5|15) 

exten => s,5,Background(pls-hold-while-try) 

exten => s,6,Goto(parkedcalls,70,1) 

exten => s,7,VoiceMail(2001@default) 

exten => s,8,Playback(Goodbye) 

exten => s,9,Hangup 

[parkedcalls] 

exten => 70,1,Answer 

exten => 70,2,SetMusicOnHold(default) 

exten => 70,3,ParkAndAnnounce(asterisk-friend:/tmp/asteriskstranger:vm-isonphone:at-followingnumber:PARKED|40|local/2001@from-internal|who-r-u,s,7) 


(*The codes in red must be in one continuous line) 

I forward all calls that were parked to my existing extension 2001 (in bold type) and direct 

voicemail for unanswered calls to the same extension as well. 

Next I create a new extension using freePBX e.g. 222. Using a softphone, I connect to 

extension 222 and unconditionally forward all calls to extension 222 to extension 111 by 

dialling *72111 

I then direct all my incoming calls from my PSTN trunk to extension 222. 


The result is, when there is an oncoming call from PSTN, the call will be sent to extension 

222 and in turn, the call will be forwarded to extension 111. 

When the call gets to extension 111, the caller will hear a message saying that, 

“unrecognised calls are not accepted and asking the caller to record his/her name after the tone” 

(or something to that effect). After the caller has complied and press the # key, the call 

will be parked. 

Extension 2001 will be called by Asterisk. When you answer the call, Asterisk will 

announce that there is a call by on extension 71 (or 72 depending on 

how many calls are being parked). If you recognize the name recorded, or if you want to 

take the call, simply hang up and dial 71. You will be connected to the caller. While this 

is all happening, the caller will be put on hold listening to your MOH. 

If however you do not wish to accept the call, you simply hang up and do not dial 71. The 

caller will be asked to leave a message instead. 


19 CONFERENCE 

Tele-conference can be configured through FreePBX Conference option. 

Conference is pretty easy to set up, as there is very little to do. However, to set up 

Conference, a Zaptel card may be required (at least in my case as I don’t seem to be able 

to get it working without a Zaptel card installed). In the latest Trixbox 2, this seems to 

have been resolved. 

The Conference function of Asterisk is similar to a Tele-conference call where multiple 

callers can call in and participate in a two-way conference like in a party room where 

everyone can talk and listen to one another or just to listen to a Tele-presentation. 

In Trixbox, there are 2 .conf files that relates to conferencing. They are: 

meetme.conf 

meetme_additional.conf 

meetme.conf 

Ensure that you have the following 2 lines (only 2 lines in the file): 

[rooms] 

#include meetme_additional.conf 

All these lines do is point to the meetme_additional.conf where the conference 

room extensions are being created when you create a conference room through 

freePBX conference setup option. 

meetme_additional.conf 

This is where the extensions for the meeting rooms are created. One line per room. 

You may have just one extension for your meeting room or you may have several 

meeting rooms. In my case, I have 1 meeting room and therefore my 

meetme_additional.conf contains the following extension: 

; Extensions for my meeting rooms 

conf => 8200| 

19.1 CREATE CONFERENCE 

FreePBX allows you to create one or more conference room through the Setup -> 

Conference screen. All you have to do is fill in the appropriate fields on the screen and 

you are on your way to a voice conference. 

Naturally there are some options that you may wish to have for the conference room. 

They are entirely up to you. The main important things are for you to create the 

conference room number and the conference name for you to know what that 

conference room is for. The rest of the fields are optional. 

Below is what my conference configuration page looks like: 


• This conference room number is 8000 

• I call this conference room General 

• Users are not required to enter a PIN number to join the conference. 

• Admin PIN number is not required for the conference to wait for the leader to 

conduct the conference. 

• I don’t have a joining message greeting when someone joins the conference. 

• I have No in the leader wait option for the conference to wait for the conference 

chairman. 

• I require a user count so I have Yes in the user count field. 

• I need to be informed when a user joins or leaves the conference. I have Yes 

here. 

• The music on hold is also flagged as Yes as I want music on while waiting. 

Once these are done, from one of your phones dial, 8000 and you will be able to join the 

conference. 

You are now in the conference room and every time someone joins or leaves the 

conference, it will also be announced. 

To test this, go to another phone and also dial 8000. You should hear an announcement 

(if you are listening on the first phone), indicating that someone has joined the 

conference. 

If there are 2 (or more) of you, you will be able to talk to each other. If one of you hangs 

up, you will know about it. 

19.2 INVALID CONFERENCE NUMBER 

If you get this error, chances are, it is caused by not having a zaptel timing source. This 

often happens when you have upgraded your kernel, but you haven't recompiled your 

zaptel to suit. 

Here is the quick fix compliment of Rob, our freePBX guru. 

Log in to your Asterisk box as root, and at the command prompt, do the following: 

cd /usr/src/zaptel 


make install 

/etc/init.d/zaptel stop 

/etc/init.d/zaptel start 


20 VOICEMAIL 

 

If you have enabled voice mail when setting up your extension, and you hear a 

stutter tone (not the normal dial tone) when you initially pick up your phone, it is 

indicating to you that you have a voice mail waiting. 

You must run setup-mail from the command line initially for Voice Mail and 

Notification to work in Trixbox. 

To enable voicemail on an extension simply "enable" it when you create the extensions 

from the freePBX GUI. If you require email notification of your voicemail, you may enter 

your email address in the email address field. 

Click submit when done followed by the red bar on top of the scree. 

20.1 ENABLING EMAIL NOTIFICATION 

Asterisk can send email to you informing you that you have voicemail waiting in your 

mailbox. In fact it can even send you the voicemail itself in the form of a .wav file if you so 

desire. 

20.1.1 Installing sendmail 

This is not always the case in every installation. If this happens to you, you may need to reinstall 

sendmail. 

I have been having some problem to get mail notification to be sent to my email 

address. I later discovered that sendmail needed to be installed for this function to 

work (thanks Thunderbird1 for pointing this out to me). Sadly, this is not mentioned 

anywhere and has caused me major grief. 

To install sendmail with yum, you can do it on the Asterisk box itself or connect 

through SSH (I use putty to connect), by logging in as root. 

Do these commands at the command prompt: 

yum install sendmail-cf 

This installs the necessary files for configuration of the sendmail client. Select “y” for 

yes when prompted and sendmail will be installed. 

Next you will need to do a little editing. I use nano to edit, thus: 

nano /etc/mail/sendmail.mc 

Locate the following line: 

dnl define(`SMART_HOST',`smtp.yourprovider') 

and edit it as per the example below 

define(`SMART_HOST',`smtp.yourprovider') 

Replace ‘smtp.yourprovider’ with your real smtp provider e.g. ‘smtp.ozemail.com.au’ 

Then press CTRL-X, and type in Y, hit enter. You'll be asked for the file name you 

want "File Name To Write: /etc/mail/sendmail.mc", and hit enter. 


Next, you will need to rebuild your /etc/mail/sendmail.cf file. 

At the command prompt, do the following: 

make -C /etc/mail 

Once done, restart sendmail thus; 

/etc/init.d/sendmail restart 

You should now be able to have your voice mails sent to you via email. 

The above instruction is also mentioned in this link: 

http://sourceforge.net/forum/message.phpmsg_id=3387743 

and http://sourceforge.net/forum/message.phpmsg_id=3432175 

20.1.2 /etc/hosts 

There is a couple of things that you will need to do as your email host may not accept 

outgoing emails unless your system has a valid DNS name. 

By default, your host will look like this: 

# Do not remove the following line, or various programs 

# that require network functionality will fail. 

127.0.0.1 trixbox1.localdomain trixbox1 localhost.localdomain localhost 

Your mail server will not accept this as a valid DNS. Therefore you will need to use 

your valid Domain Name e.g. pbx.myasterisk.com. If you are like me, without a 

valid Domain Name, you should obtain one through DynDNS. 

Using Nano, you will need to edit your /etc/hosts file from command line to this: 

# Do not remove the following line, or various programs 

# that require network functionality will fail. 

127.0.0.1 localhost 

127.0.0.1 pbx.myasterisk.com (use a valid domain name here) 

127.0.0.1 trixbox1.localdomain trixbox1 localhost.localdomain 

You may use an editor (such as nano) to edit /etc/hosts file through CLI. 

 

If you are using SipBroker or Pennytel, its best not to make this change 

as describe above, as SipBroker and Pennytel will resolve your DNS name 

as the local host address of 127.0.0.1 as defined in the hosts file and not 

your actual external IP address. To overcome this I substitute my DNS 

hostname with another valid DNS name such as yahoo.com (as I have an 

account with Yahoo – not sure about the legality side of this though). 

Refer to Notes for a kludge fix if you are experiencing difficulties. 

Next, if you wish, you will need to customize your email notification message in 

/etc/asterisk/vm_email.inc file and to include the actual IP address of your asterisk 

PC. I have found that this is best done through a text editor (e.g. Nano) as AMP 

tends loose the formatting character. In the later versions of Trixbox, this can be 

done through Config Edit. 


; Change the email body, variables: VM_NAME, VM_DUR, VM_MSGNUM, 

VM_MAILBOX, VM_CALLERID, VM_DATE 

emailbody=${VM_NAME},\n\nThere is a new voicemail in mailbox 

${VM_MAILBOX}:\n\n\tFrom:\t${VM_CALLERID}\n\tLength:\t${VM_DUR} 

seconds\n\tDate:\t${VM_DATE}\n\nDial *98 to access your 

voicemail by phone.\nVisit http://pbx.myhome.com:8080/cgibin/vmail.cgiaction=login&mailbox=${VM_MAILBOX} 

to check your 

voicemail with a web browser.\n 

You can be aggressive and change /etc/asterisk/vm_general.inc as well. 

; 1st listed format gets emailed 

format=wav49|wav 

attach=yes 

pbxskip=yes 

; Don't put [PBX]: in the subject line 

serveremail=vm@asterisk ; Who the e-mail notification should appear to come from 

fromstring=Voicemail System ; Real name of email sender 

maxmessage=180 

; max length of vm message 

minmessage=3 

; Minimum length of a voicemail message in seconds 

maxsilence=5 

; Wait for 5 silent seconds and end the voicemail 

silencethreshold=128 

; What do we consider to be silence 

skipms=3000 ; ; How many millisecs to skip forward/back when rew/ff in 

; message playback 

review=yes 

; Allow sender to review/rerecord their message 

; before saving it 

operator=yes ; Allow caller to press 0 

nextaftercmd=yes 

; Automagically play next message when current 

; message is deleted or saved 

Change the entry in red to reflect your requirement e.g you may change vm@asterisk 

to FHC PBX and change Voice mail System to read Family Health Centre Voicemail. 

Once these are done, you will receive email notification of voicemail left in your 

voicemail box. 

You may start testing it and your email received will look something like this: 

From: Family Health Centre Voicemail To: Ben Sharif 

Subject: New Message 1 in mailbox 201 

Ben Sharif, 

There is a new voicemail in mailbox 201: 

From: "Mary" 

Length: 0:20 seconds 

Date: Friday, July 29, 2005 at 11:16:38 PM 

Dial *98 to access your voicemail by phone. 

Visit http://pbx.familyhealth.com:8080/cgibin/vmail.cgiaction=login&mailbox=201 

to check your voicemail with a web browser. 


20.2 USE POSTFIX INSTEAD OF SENDMAIL 

Another alternative to email notification is to use Postfix instead of Sendmail because 

Sendmail is not as secure besides, using Postfix is a lot easier and it is more secure than 

Sendmail. 

If you wish to gi this path, it is strongly recommend you remove Sendmail and replace it 

with Postfix, with only 2 easy commands: 

rpm -e --nodeps sendmail 

yum -y install postfix 

Following this, edit your /etc/postfix/main.cf, (use nano and my smtp server is shown 

below, but you should be using yours) and add: 

relayhost = smtp.ozemail.com.au 

Replace smtp.ozemail.com.au with your own smtp server. E.g. if your belong to OPTUS, 

it should be: 

relayhost = mail.optusnet.com.au 

You should also edit /etc/asterisk/vm_general.inc and set the "serveremail" parameter to 

your real email address. This is to avoid messages with invalid from-addresses floating 

aimlessly in the Internet. 

Then load Postfix. 

service postfix reload 

Postfix will now send all outgoing email to your ISP's mail server. 

20.3 HOW TO RETRIEVE VOICEMAIL FROM EXTERNAL PHONE 

To call in and retrieve your voicemail from external phone is as easy as simply calling 

your number and retrieve your voice mail. This can be done easily if you have IVR 

enabled in your new TRIXBOX system. 

This is how I do it (you may have been told of other different methods). 

20.3.1 Method 1 

1. Call my number and when the Digital Receptionist answers, I will dial my 

extension number prefixing it with an *. E.g. My extension number is 2001, 

so when the DR answers me, I will dial *2001 

2. I will then be ask to leave a voicemail, but instead, I will dial * again. 

3. DR will ask for a password to be followed by the # key. 

4. I enter my password followed by # 

5. DR will then give me my voice mail if there is any. 


20.3.2 Method 2 

1. From freePBX Setup Screen, create a Misc Destination and give it a name 

of e.g. “Check VM” 

2. Select Feature Code Dial Voicemail (*98) from the dropdown menu. 

3. Submit the changes. 

Next modify your IVR 

1. Increase option. If you already have 7 options, this will be the 8 th . 

2. Create an unannounced IVR selection e.g. selection no: 8 

3. For this option, select Misc Destination and from the drop down, select 

“Check VM” 

4. Save and don’t forget to click on the red bar. 

5. When you call your PBX from an external phone, DR will naturally ask you to 

make a selection. Selection 8 will not be announced, however you should 

select 8. 

6. DR will ask for a mailbox number 

7. Enter your extension number (which is also your mailbox number) 

8. Dr will then ask for your password 

9. And you will get your mailbox IVR prompt. 

As I said, it is indeed simple. 

20.4 HOW TO CHANGE VOICE MAIL GREETINGS 

The default Voicemail greeting is fine but it is rather bland and quite boring. No doubt 

some user would like to change the standard greeting to something more personal and 

meaningful. 

 

Ensure that you have changed your Direct Dial to Voicemail message type, in 

General Setting, from Default to something else e.g. Unavailable. Otherwise 

when you use follow-me and the like, your custom recorded message will not 

be played back. The Asterisk default message will be played back instead. 

To do so is pretty simple. Here are the steps: 

1. Use your existing extension and dial *97 

2. You will be asked for your password (if you have entered one) 

3. When the Voicemail IVR starts, press 0 

4. You will then be given the choice what type of message you want to record. 

5. Choose the appropriate message you want to record. 

6. Record your message “Thank you for calling, I am either busy elsewhere or on the 

phone. Please leave me your message and I will call you back as soon as I can”. 

7. Review your message 

8. Re-record if you want to or if you are happy with the message, choose the option 

to save. 

That’s all – you are done. 


21 FAX TO E-MAIL 

Before setting up Fax over IP, I suggest that we should read this: http://www.softswitch.org/foip.html 

and : http://www.voip-info.org/wiki-Asterisk+fax 

While sending and/or receiving Faxes is one of the available functions in TRIXBOX, there 

have been reports of various degree of success. This is very much dependant on the 

type of equipment used such as PC, processor speed, line quality etc. 

In the earlier incarnation of TRIXBOX (it was AAH for those old enough to remember), for 

the fax facility to work reliably, you will also need Newman Telecom’s NVfaxdetect. This 

can be downloaded FREE. This may require some modification and recompiling. 

SpanDSP allows Asterisk to receive faxes by converting the received data to .pdf format. 

This service must be installed. If you want to have this function, you will need to install the 

support for encoding incoming faxes as PDF. This support is not pre-installed with 

TRIXBOX although the necessary files are there for you to manually install. 

21.1 INSTALLING FAX FOR TRIXBOX 

Unlike previous versions of Trixbox/AAH where it has to be installed separately, the later 

versions of Trixbox uses Asterisk version 1.2.2 where the developers of Asterisk have 

included NVfaxdetect in Asterisk which allows incoming faxes to be received over SIP, 

IAX & ZAP channel.. 

There has been mixed reactions on the ability of sending and receiving fax through 

Trixbox as such I would suggest that, if you are relying on fax, to use a dedicated fax 

machine and only use Trixbox for experimental purposes only (at this stage anyway). 

 

Whatever it is worth, I have not been able to utilise this facility to my 

satisfaction. 


22 WEATHER REPORTS 

The existing weather forecast information that comes with TRIXBOX is configured for 

New York. However, Asterisk can be configured to allow the automatic downloading of 

Australian weather forecast information, convert it to a .wav file and play the 

announcement. The announcement will be a simulated announcement (ala Dr. Who’s 

Dalek). However it will provide you with the weather report of your region of choice. 

The Australia weather report reference file can be obtained from the following site: 

http://www.bom.gov.au/ 

You may choose the reference file (text file) for the area that you are interested in. You 

do not need to manually download the file, instead only the file name is required to be 

noted and can be incorporated within the Trixbox’s extensions_trixbox.conf file later. 

The script will download the file for you in the background. 

The following are examples of available weather report files that you can reference in 

your script. 

Forecast Area 

NSW Sydney Metro 

All NSW 

ACT 

Snowy Mountain 

Brisbane Metro 

All Queensland 

Melbourne Metro 

All Victoria 

Adelaide Metro 

All SA 

Perth Metro 

All WA 

Darwin and Rural Area (All NT) 

Tasmanian Cities 

All Tasmania 

File reference 

IDN10064 

IDN10031 

IDN10035 

IDN10103 

IDQ10090 

IDQ10700 

IDV10450 

IDV10310 

IDS10034 

IDS10033 

IDW12300 

IDW12000 

IDD10110 

IDT13400 

IDT13100 

For a complete list, refer to the link below: 

http://www.bom.gov.au/inside/cosb/stcm/idcodes/tables/current/vic_products.shtml 

The following step-by-step instruction will guide you through. 

22.1 CONFIGURE WEATHER REPORT – ON DEMAND. 

Some preparations need to be manually conducted through Linux Command Line, or if 

you prefer, you may use Webmin. 

I used Putty to log on to the Asterisk box remotely and used Nano to create a text file 

called “convert2wav” in the /usr/bin directory containing the following: 

/usr/bin/flite /tmp/weather.txt /tmp/weather.wav 

(This command is used to convert the weather text file to a .wav sound file). 

Once this is done, the permission for the “convert2wav” text file need to be appropriately 

set to allow it to be executed. To do this you need to run the following command from 

Command Line. 

chmod 755 /usr/bin/convert2wav 


Now you have completed all the necessary preparations – for me anyway. 

22.1.1 Create Extension codes. 

The next step is to include the following in the extensions_trixbox.conf through the 

Config Edit option of TRIXBOX Maintenance Tab. Select an unused extension code 

e.g. *61, *63, *64 etc. 

Add the following example (*Note: The third line in each section (bold) should read as a 

complete single line- change the file reference in red to suit.). 

; ################################## 

; Forecast for Sydney Metro 

; ################################### 

exten => *61,1,Answer 

exten => *61,2,Playback(national-weather-service) 

exten => *61,3,Playback(pls-hold-while-try) 

exten => *61,4,System(/usr/bin/wget -O weather.txt 

ftp://ftp2.bom.gov.au/anon/gen/fwo/IDN10064.txt) 

exten => *61,5,Wait(1) 

exten => *61,6,System(/usr/bin/convert2wav) 

exten => *61,7,Playback(/tmp/weather) 

exten => *61,8,System(rm /tmp/weather.* -f) 

exten => *61,9,Hangup 

; ###################################### 

; Forecast for the whole of NSW 

; ###################################### 




exten => *63,4,System(/usr/bin/wget -O weather.txt 

ftp://ftp2.bom.gov.au/anon/gen/fwo/IDN10031.txt) 

exten => *63,5,Wait(1) 

exten => *63,6,System(/usr/bin/convert2wav) 

exten => *63,7,Playback(/tmp/weather) 

exten => *63,8,System(rm /tmp/weather.* -f) 


Restart asterisk and test. You will be presented with the weather report of the areas 

you have chosen. 

(Source Reference: 

http://forums.whirlpool.net.au/forum-replies.cfmt=364420&p=1) 

This method is very slow and therefore is only suitable for a very fast PC. Depending on 

the speed of your computer during the .wav conversion, there may be about 1 minute or 

more delay between dialling the extension and the weather announcement. Therefore I 

have included step 3 telling the caller to hold while trying to connect. 

Therefore, there has to be a better method – so read on. 

22.2 CONFIGURE WEATHER REPORT – BACKGROUND METHOD. 

After giving it a bit of thought, I have decided to let Asterisk do the task of getting the 

weather text files and converting them to the necessary sound files in the background 

every 4 hours. This will cut down the waiting period when a caller chooses to listen to the 

weather report. 


First we will use putty to make a batch text file called “convert2wav” in the /usr/bin 

directory containing the following: 

/usr/bin/wget -O /tmp/weather1.txt ftp://ftp2.bom.gov.au/anon/gen/fwo/IDN10064.txt 

/usr/bin/flite /tmp/weather1.txt /tmp/weather1.wav 

/usr/bin/wget -O /tmp/weather2.txt ftp://ftp2.bom.gov.au/anon/gen/fwo/IDN10031.txt 

/usr/bin/flite /tmp/weather2.txt /tmp/weather2.wav 

(Change IDN10064.txt and IDN10031.txt to suit the areas you require) 

If you want to get rid of a couple of nasty characters, you can always add the following 

line or lines in the appropriate places and changing the characters that you want to 

discard or change, 

# sed 's/\[/ /g' /tmp/weather1.txt > temp.txt 

# sed 's/\]/ /g' /tmp/weather1.txt > temp.txt 

# sed 's/\_/ /g' /tmp/weather1.txt > temp.txt 

as an example 

- but we are not going there. It may get too confusing. For the moment we will just stick 

with the batch file example above. 

Once this is done, change the permission for “convert2wav” thus: 

chmod 755 /usr/bin/convert2wav 

Use Webmin to schedule a Cron job to run this batch file every 4 hours (or schedule it to 

suit your purpose). This will run the batch job at the scheduled time avoiding having to 

download the weather file and converting them to sound files when a caller chooses the 

weather report option. 

The sound files will always be available when a caller requires them without having to 

wait for them to be downloaded and converted. 

22.2.1 Create Extension codes. 

Next include the following in extensions_trixbox.conf. Select an unused extension 

code e.g. *61, *63, *64 etc. 

Add the following example 

; ################################## 


; ################################### 




exten => *61,4,Playback(/tmp/weather1) 


; ###################################### 


; ###################################### 




exten => *63,4,Playback(/tmp/weather2) 



Restart asterisk and test. You will be presented with the weather report of the areas 

you have chosen without any delay whatsoever. 

22.3 NOW PUT IT IN YOUR DIGITAL RECEPTIONIST 

If you want the 2 procedures to be incorporated in your Digital receptionist, simply 

duplicate them in your extensions_custom.conf with some minor changes. You will 

want the digital receptionist to know which weather report you are referring to when you 

make the choice. Give it a label – see example below (the changes made are in bold). 

; ################################## 


; ################################### 

[custom-weathermetro] 


exten => s,2,Playback(national-weather-service) 

exten => s,3,Playback(pls-hold-while-try) 

exten => s,4,Playback(/tmp/weather1) 


; ###################################### 


; ###################################### 

[custom-weathernsw] 


exten => s,2,Playback(national-weather-service) 

exten => s,3,Playback(pls-hold-while-try) 

exten => s,4,Playback(/tmp/weather2) 


When you create your menu option in Digital receptionist, e.g. “Press 6 for Sydney 

weather report, press 7 for the whole of NSW weather report”, select the custom app: and 

enter the appropriate custom procedures above. See example below: 

Custom App 

Custom App 

custom-weathermetro,s,1 

custom-weathernsw,s,1 


23 REMOTE MANAGEMENT 

Some may want to manage their Asterisk Box remotely. If you are thinking about doing 

this, I highly recommend you register with Dyndns to get yourself a dynamic DNS unless 

you have a fixed public IP address. 

You should also forward port 80 on your router to point to the local IP address of your 

Asterisk box. 

To log on to your Asterisk box remotely using your browser, use 

http://your_Dyndns_name:80 or http://your_public_ip_address:80 

23.1 HTTPD.CONF 

If you are using Optus Cable, you may not be able to use port 80. However this can be 

overcame by changing the port to 8080. To do this, you will be required to change one 

setting in the httpd.conf file which resides in the /etc/httpd/conf directory. 

If you are not a Linux fan, I suggest you use either webmin or putty to access the file. I 

use Putty to log on remotely to the console and edit the file with Nano using the following 

command. 

nano /etc/httpd/conf/httpd.conf 

In the file, look for the following entry (marked in red) - below : 

# 

# Listen: Allows you to bind Apache to specific IP addresses and/or 

# ports, in addition to the default. See also the 

# directive. 

# 

# Change this to Listen on specific IP addresses as shown below to 

# prevent Apache from glomming onto all bound IP addresses (0.0.0.0) 

# 

Listen 80 change this from 80 to read 8080 

# 

After doing the above, reboot Asterisk. 

Don’t panic if you can no longer access AMP through you web browser. You simply 

append: 8080 after the address e.g. 192.168.1.101:8080 


24 HOW TO SET UP A REMOTE EXTENSION 

If you carry your laptop/notebook everywhere you go, you may want to consider setting 

up a remote extension on your Notebook. Basically its set up is similar to the softphone 

that you set up on your other computers except that you should have a fixed IP address 

for your Asterisk server or use DynDNS to register a Domain ID for your Asterisk box. 

Since I am on Optusnet cable, I only have the privilege of a dynamic IP hence I registered 

for a Domain ID with DynDNS pointing to my Dynamic IP address. Every time my 

dynamic IP changes, DynDNS will point to the new IP address. This way I don’t have to 

keep changing IP addresses. I simply use the same DNS ID eg: myasterisk.freedns.com 

24.1 CREATE A NEW EXTENSION 

Create a new IAX2 extension in Asterisk – just like the other extension that you created 

before. This time instead of creating a SIP extension, create an IAX extension (you can 

use SIP as well but since you don’t have any control of the external router that you may 

be connected to when you are away from your own environment, IAX will avoid any NAT 

issue that you may encounter). In your router, you will need to forward port 4569 to your 

Asterisk Box. 

As mentioned above, you may create a SIP extension instead, if you so wish but you 

must ensure that you set “nat” field to “yes” ( in the device option) instead of “never” as 

illustrated in the screen below - see red arrow. 

Once you have done this, you will need to ensure a couple of things in your .conf files are 

set correctly. 


24.2 IAX.CONF 

The following is the content of my iax.conf file. 

iax.conf 

[general] 

bindport = 4569 ; Port to bind to (IAX is 4569. Forward this port to your Asterisk box) 

externip = or 

externhost = 

localnet = 192.168.1.0/255.255.255.0 

bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) 

delayreject = yes 

disallow=all 

allow=g729 

allow=ilbc 

allow=gsm 

allow=g723 

allow=alaw 

allow=ulaw 

jitterbuffer = yes 

mailboxdetail = yes 

#include iax_additional.conf 

#include iax_custom.conf 

24.3 SIP_NAT.CONF 

The only other .conf file that requires attention is the sip_nat.conf in case you want to 

use SIP. If you do, ensure that you have forward ports 5004 – 5082 as well. My file is as 

per the example below. 

sip_nat.conf 

nat=yes 

externip= or 

externhost=< myasterisk.freedns.com> (see note) 

localnet=192.168.1.0/255.255.255.0 

externrefresh=10 

Those are the only 2 .conf files that require changing if needed. 

24.4 SETTING UP AN IAX SOFTPHONE 

I use an IAX softphone called IDEFISK. You can get IDEFISK from here: 

http://www.asteriskguru.com/tools/idefisk_beta.php 

You can also use Virbiage softphone as it will handle IAX. 

Setting up IDEFISK is very simple. All you have to do is download the file, extract it and 

save it in its own directory. 

Click on the idefisk.exe and just configure it like the example below. 


Once the softphone is active, to start 

configuring, simply place your curser 

anywhere in the middle if the image and 

right click. 

The configuration option window will be 

displayed. 

24.4.1 First the general options 

In the general options, checked the 

“Check for update” and “Automatic 

popup on incoming call” options. 

The next option to take care of is the Account option. Apart from that, there is nothing 

else critical to be done for you to use IDEFISK. 


24.4.2 Accounts options 

Next click on the Account Options and set up your account as per the image below. 

The rest of the options are there for you to set up your sound preference and customize 

your look and feel. They are pretty straightforward. 

Now you are ready to go on the road with an Asterisk extension under your arm. 


25 CALLS RECORDINGS 

There have been numerous discussions on the legality of recording telephone 

conversations. Some say its OK while others say its illegal. If you are recording 

conversation without the consent of all parties, you may be breaking the law – so be 

warned. 

For those who do record conversations, the recording are kept in the 

/var/spool/asterisk/monitor directory or you can monitor the recording through the ARI 

interface. 

25.1 ARI INTERFACE 

The ARI interface can be access either from the User Mode screen of Trixbox or freePBX 

GUI. 

Use mode interface of Trixbox 

FreePBX GUI of Trixbox 

If you want to monitor all the calls, you will need to log in as admin with the default 

password of ari_password. 

To monitor the calls for any individual extension, log in using the extension number and 

the extension password. 


To play the recording, simply click on the Play link related to the extension that you want 

to listen to, in the screen that follows (ensure that you have installed Quicktime and Adobe 

Flash Player plugin for your browser on your local PC, otherwise it will not play the sound). 

The configuration file for ARI is /var/www/html/recordings/includes/main.conf.php 

There are 2 ways you can record your calls: 

25.1 RECORD ALL CALLS BY DEFAULT 

To record all calls, on a specific extensions or all extensions, this function has to be 

enabled on the extension or extensions involved. This is done through the Extension 

Options Record Incoming and Record Outgoing field by selecting “Always” from the 

dropdown box for both fields. 

25.2 RECORD CALLS ON DEMAND. 

To record on demand, the caller or the receiver have to dial *1 when he/she need the 

recording to start (Asterisk v1.2). This function will only work if the feature is enabled. 

To enabled this feature, the following Dial Command Option must be enabled in freePBX 

General settings Asterisk Dial command option and Asterisk Outbound Dial 

command option fields by adding the following flag in the fields. 

w: Allow the called user to start recording after pressing *1 (Asterisk v1.2) 

W: Allow the calling user to start recording after pressing *1 (Asterisk v1.2) 

Some users have noticed that the *1 need to be pressed very quickly for this feature to 

work. This can be remedied by adding the following line in the features,conf file under 

the general section. 

featuredigittimeout = 1500 ;(the default is 500 which is too fast for an old guy like me 


26 STUN SERVERS 

STUN (Simple Traversal of UDP through NAT - Network Address Translation) is a 

protocol for assisting devices behind a NAT firewall or router with their packet routing 

operates on TCP and UDP port 3478. 

I never have the need to use STUN. The list below is provided for those who have the 

need to use public STUN Servers. These are the servers that I am aware off - there may 

be others. 

stun.xten.com 

stun.fwd.org 

stun01.sipphone.com 

stun1.noc.ams-ix.net 

stun.voipbuster.com 

stun.voxgratia.org 

27 DIALING THROUGH MS OUTLOOK® 

One way of keeping a little black book for all your telephone contacts is through 

Microsoft Outlook® (for the windows users) address book. We can also use Outlook to 

dial the phone numbers in the address book as Outlook® has a click-to-dial facility. 

I was alerted of this facility in Outlook® and Asterisk® and asked to prove the application 

and publish a simplified instruction in this guide. 

27.1 DOWNLOAD ASTTAPI 

Before you get started, you will need to download AstTapi, which is a Microsoft TAPI to 

Asterisk Bridge. This little gem (still under further development at the moment but 

functional) will make it possible to do click-to-dial from Microsoft Outlook and for that 

matter from other TAPI compliant applications. 

Download AstTapi from http://sourceforge.net/projects/asttapi/ 

To get started, you will need to edit manager_custom.conf file. The easiest is to use 

freePBX Config Edit (the easiest way!) 

SETUP -> Config Edit -> manager_custom.conf 

On opening the manager_custom.conf file, you will notice the following, which you will 

need to do a tiny weeny edit: 

[phpconfig] 

secret = php[onfig 

deny=0.0.0.0/0.0.0.0 

permit=127.0.0.1/255.255.255.0 

read = system,call,log,verbose,command,agent,user 

write = system,call,log,verbose,command,agent,user 

[phpagi] 

secret = phpagi 

deny=0.0.0.0/0.0.0.0 

permit=127.0.0.1/255.255.255.0 



[AstTapi] 

secret = AstTapi 

deny=0.0.0.0/0.0.0.0 


#permit=192.168.1.0/255.255.255.0 



Under the [AstTapi] User context, remove the # from the permit line and change the 

192.168.1.0 to the network address of your network. In my case I left it as it is because 

my network is under 192.168.1.0 (see red highlight). 

Once this is done, click Update and restart Asterisk. 

27.2 INSTALL ASTTAPI 

Install AstTapi on your PC – the one that you have Microsoft Outlook installed. Follow the 

prompt and once completed you will need to restart your PC for the new driver to take 

effect. 

27.3 CONFIGURE OUTLOOK ADDRESS BOOK 

Fire up Outlook®. 

Click on Contacts from your 

folder list. 

Select a contact that you 

want to call (provided that a 

telephone number has been 

recorded). 

Click on the phone icon. 

Click on the Telephone icon. 

The New Call dialog box will 

be displayed: 

Select Dialling Options. 


The Dialling Option box will 

be displayed. 

In the "connect using line" 

drop down box, select 

Asterisk. 

Then click on Line 

properties. 

In the Host field, enter the 

address of your asterisk 

server: e.g. 192.168.1.100 

In the Port Field: 5038 

User Field: AstTapi 

Password: AstTapi 

User Channel: (your extension – the extension you want Asterisk to ring before 

connecting you the number you are calling) e.g. sip/2001 

Select dial by context radio button. 

Enter outbound-allroutes in the context box 

When done, click Apply and you are done. 

You can now start making calls from Outlook using the click-to-dial facility. 

Now… that was quite painless. 


28 SPEED DIAL 

Sometime you may just need to call someone quickly without having to look up his/her 

phone number. Well, you can by simply defining a shortcut number. There are 2 ways 

you can do this - my way and Trixbox’s way. 

28.1 THE DUMB-ME METHOD 

Since I am very bad at remembering phone numbers – I cant remember my own phone 

number (or my car registration number), I have defined some shortcut numbers for my 

family using the first 3 characters of their first names e.g. Adam is ADA which translate to 

232 on the telephone keypad. Make sure the numbers have not been allocated to your 

local extension. 

To do this, you will need to edit extensions_custom.conf by adding the lines below 

containing the relevant exten expressions. 

You can do this through Setup -> Config Edit 

; ############################################################# 

; Speed Dial to some predefined numbers (added by Ben 2/12/2005) 

; ############################################################# 

exten => 236,1,Dial(SIP/6140036xxxx@pennytelsip,60,m) ; Ben’s Mobile 

exten => 236,n,Hangup 

exten => 764,1,Dial(SIP/6141242xxxx@pennytelsip,60,m) ; Rohani’s Mobile 


exten => 667,1,Dial(SIP/6140470xxxx@pennytelsip,60,m) ; Nor’s Mobile 


exten => 767,1,Dial(SIP/6140512xxxx@pennytelsip,60,m) ; Ros’s Mobile 


exten => 232,1,Dial(SIP/6140936xxxx@pennytelsip,60,m) ; Adam’s Mobile 


You will need to substitute the example phone numbers above such as 6140036xxxx with 

the real phone numbers. 

When 236 (which is BEN) is dialled, you will hear a voice prompt saying “Please wait 

while I connect your call” followed by Asterisk calling Ben’s Mobile number which is 

61400336510 using the Pennytel SIP trunk. 

Pennytel SIP trunk was called Pennytelsip when it was initially set up. 

The call will be tried for 60 seconds (,60) and while the call is being attempted, the caller 

will be listening to a MOH tune (,m) instead of the ringing tone. You may however 

change it to a ringing tone by substituting the “m” with a “r” or remove “,m” altogether 

Once you have done that, click on Update and re-read the config. That’s all there is to it. 


28.2 TRIXBOX’S METHOD (VERSION 1.X) 

In Trixbox v1.x, there is however, a simpler way (Later versions have different method) 

which I will cover after this. 

TRIXBOX has included a built-in speed dialling utility amongst the goodies that comes 

with it. Numbers 300 to 399 has been reserved for this purpose. 

To add a number to your speed dial list, you simply pick up one of your extensions and 

punch in 3003XX61212345680 (assuming 61212345680 is the number you want to save 

in your speed dial list) where 3XX is the speed dial code for 61212345680 (substitute 3XX 

for the position e.g. 301). 

After doing that, Asterisk will confirm the code and the number that you have just 

registered. If you want to make correction to or change the telephone number that you 

have entered simply repeat the process. 

In the above example, every time you dial 301 Asterisk will dial out the number 

associated with that code. In this case, it is 61212345680. 

Ensure that you enter the number to be called in the same format that is supported by 

your Asterisk dialplan. 

To look up speed dial numbers, simply dial * followed by the 3-digit speed dial code, e.g. 

*301. This will tell you the number stored in speed dial position 301. 

28.3 TRIXBOX’S METHOD (VERSION 2.X) 

In Trixbox v2.x, the method in version 1.x has been revised to use the Asterisk Phone 

Book found under the TOOLS Tab. 

Here you simply enter the Phone number of the party you wish to call, the name and the 

Speed Dial Code in the appropriate fields as per the example below. Submit change 

when done. 

To call the Speed dial, e.g. you want to call Ben Sharif, simply dial *069 (the *0 tells 

Trixbox that you want to use the Speed Dial and the 69 is the Speed Dial Code for Ben 

Sharif). 


The Speed Dial Prefix *0 can be found under Feature Codes. 

Naturally if you have a big list of Names and Phone Numbers that you want to enter in 

your Trixbox Phone Book, you have the option of uploading the list. You have to scroll 

down the page to get to the “Import from CSV” section of the Phone Book (see illustration 

below) 

. 

The “Import from CSV” is a misnomer because, if you havae created a list using EXCEL 

and trying to import the list, it will not work since EXCEL CSV file is a comma delimited 

one where as freePBX is expecting a semi-colon delimited file. Mousing over the word 

file will give you the format it is expecting; 

“Name”;Number;Speeddial code 

Ensure that “Name” must be enclosed in quotation marks. 

Any speeddial code that already exist in the phone book cannot be reused. If you want to 

use an existing speeddial code in your so called “CSV” file, the existing entry in the phone 

book must first be removed. 


29 HOW TO USE WINDOW MESSENGER® 5.X 

 

Windows Messenger is NOT MSN Messenger. You may need to download 

Windows Messenger from Microsoft. 

Set up Windows Messenger 5.x as a SIP client on your PC: 

1. Start Windows Messenger from your PC. 

From the Windows Messenger dialog box, go to Tools -> Options ->Accounts: 

• Check the [My contacts include users of a SIP Communications Service] checkbox. 

• Enter "sip:@" 

(e.g.."sip:206@192.168.1.101") in the [Sign-in name] field 

2. Click on the [Advanced] button next to the Sign-in name. 

• Check the [Configure settings] button 

• Enter Asterisk IP address (e.g.192.168.1.101) in the [Server name of IP address] 

field 

• Check the [UDP] button 

• Click [OK] and [OK] again. 

In order for Windows Messenger to work as a voice communication device, you must 

have a microphone and speaker(s) connected to your PC. 


This setting means the Windows Messenger has a user/phone number of 206 

known to the SIP server (your Asterisk box) of 192.168.1.101. 

Verify that Windows Messenger has successfully registered with Asterisk by logging in to 

Asterisk. 

freePBX -> Tools -> Asterisk Information. – this is available as an addon (see NOTES 

on Adding the missing Maintenance Modules) 

You should see under SIP Registry that the Windows Messenger is registered as a 

user/phone number of 206, with a contact URL of your local PC IP Address e.g. 

192.168.0.11, among other registration information. 

Note: Windows Messenger 5.0 or later has support for VoIP communications but it seems 

that it may create problems with chat or other features. For this purpose, please focus 

your testing on VoIP only. 


30 HOW TO INTERCONNECT 2 BOXES 

There may be a time when you want to interconnect 2 Asterisks boxes (def.com.au and 

xyz.com.au) together and if you are like me, you will probably be spending a good part of 

3 hours trying to get them to talk to one another. 

I have 2 different locations, the Main Office (def.com.au) with about 11 extensions and 

another office in a different location (xyz.com.au) about 20 km away with 9 extensions. 

The main office is the only box that will have accounts with different VSPs and all external 

communications are through the main office Asterisk box. I settled for the simplest 

solution and after some fiddling around I managed to get them to work the way I wanted it 

but not happy with it, I solicited some advise from a friend (thanks to Mark Brooker) who 

provided further assistance. 

Instead of being verbose in my explanation, I will just create a few tables outlining what I 

did. I hope this will help those in the same position as I am, to set 2 very basic systems 

together (you can refer to DUNDi for a more complete solution). A little tutorial on DUNDi 

can be found here. 

30.1 METHOD 1 - WITH THE PEER ASTERISK BOX AS EXTENSIONS 

For the purpose of registering the peers to each other, I created 1 extension on each box 

eg: 90000 on System 1 and 91000 on System 2– using extension numbers that I am not 

likely to use as local extensions (while some users have had success using common 

extension, but I prefer 2 separate extensions as I have them working). For simplicity, I 

gave a common password xxxyyy to both boxes. Avoid using extension starting with 8 as 

it may clash with conferencing. 

System 1 

def.com.au 

System 2 

xyz.com.au 

IAX Trunk 

Outgoing Dial Rules: XX. XX. 

Trunk Name Parramatta MainOffice 

Peer Details 

host=xyz.com.au (or System 2 IP) 

qualify=yes 

secret=xxxyyy 

type=peer 

username=91000 

host=def.com.au (or System 1 IP) 

qualify=yes 

secret=xxxyyy 

type=peer 

username=90000 

User Context Leave blank Leave blank 

User Details Leave blank Leave blank 

Register String 91000:xxxyyy@xyz.com.au 90000:xxxyyy@def.com.au 

Note: Registration isn’t really necessary. It will still work without it unless you use Dynamic IP. 

System 1 System 2 

Extensions 

Phone Protocol IAX IAX 

Extension Number 90000 91000 

Extension Password xxxyyy xxxyyy 

Fullname Parramatta Main Office 

Voicemail & Directory Disabled Disabled 



Outbound Routing 

Route Name Parramatta MainOffice 

Route Password Leave Blank Leave Blank 

Dial Patterns 

6XXX 

(6001 to 6009 are Parramatta 

Office extensions) 

XX. 

(Apart from Local extensions, all 

others go via Main Office) 

Trunk Sequence IAX2/Parramatta IAX2/MainOffice 

The above Outbound Routing rule assumes that you do not wish to use a dialling prefix. If 

you want to use a prefix to dial the remote extensions and to use the remote routing rules, 

you may place a prefix e.g. 9|6XXX and 9|XX. for system 1 and system 2 respectively 

instead of just 6XXX and XX. 

 

While this method will provide some rudimentary security (though pretty 

weak), as it requires an extension to be created for the peer Asterisk box, it will 

not pass the calling party extension number to the remote Asterisk box. 

Instead, it will pass the Trunk ID only and all calls will seem to come from the 

same trunk and not individual extension – I did say that this is a simple 

solution. 

30.2 METHOD 2 - IN A PEER/USER ARRANGEMENT 

Another method that I use is described below. This method treats both the Asterisk box 

as internal to each other as peer and user. I am using IAX2 for this purpose, however I 

believe, you may be able to do this with SIP as well if you are trying to connect the older 

Asterisk with the newer incarnations (I have not proved it yet). This method does not 

require registration either and does not require you to create extensions for the peers. In 

many ways, this is simpler to set up. 

Unlike the first method, this second method will pass the Caller ID to the receiving party. 

The receiving party will actually get the callers’ extension number/ID instead of the 

extension number of the peer Asterisk box. 

Rather than being verbose, I will illustrate this method using tables as follows: 

System 1 

def.com.au 

System 2 

xyz.com.au 

IAX2 Trunk 


Trunk Name InterOffice InterOffice 

Peer Details 


Qualify=yes 

type=peer 

User Context InterOffice-In InterOffice-In 

User Details 

context=from-internal 

host=xyz.com.au (or IP) 

type=user 


Qualify=yes 

type=peer 


host=def.com.au (or IP) 

type=user 




Route Name InterOffice InterOffice 


Dial Patterns 

6XXX 



9|XX. (To call external number 

via Main Office) 

XX. 



Trunk Sequence IAX2/InterOffice IAX2/InterOffice 

30.3 METHOD 3 – FRIEND/FRIEND ARRANGEMENT 

(By far the simplest method) 

This method is by far the simplest method where the boxes are treated as friend to each 

other. Like Method 2, registration is not required and this method will also pass the Caller 

ID to the receiving party. 

System 1 

def.com.au 

System 2 

xyz.com.au 

IAX2 Trunk 


Trunk Name InterOffice InterOffice 

Peer Details 

User Context 

User Details 


Qualify=yes 

type=friend 


Qualify=yes 

type=friend 



Route Name InterOffice InterOffice 


Dial Patterns 

6XXX 



9|XX. (To call external number 

via Main Office) 

XX. 



Trunk Sequence IAX2/InterOffice IAX2/InterOffice 

All the above examples assume that both Asterisk boxes have Public Fix IP address. If 

you have Dynamic IP addresses, you will need to register both the boxes with DynDns to 

obtain a valid DNS ID. If you are a part of a Corporate LAN, then you will have no need 

to worry about DynDns and what not. 

Warning: You must provide for security, as this is pretty wide open. 

Like all installation, you must provide for security. As different installation resorts to 

different types of security arrangement, I will leave that to the individual implementer to 

deal with the security issues. 


30.4 THINKING OF MORE THAN 2 BOXES 

Just as a matter of interest, you can connect several boxes using this method. While I 

have connected 3 boxes successfully, I believe, the same principle can be applied to 

more boxes. 

 

In my implementation I have box A, B and C (System 1, 2 and 3) in a star 

configuration. Box A is the master box. All the other boxes use box A as the 

main exchange. 

A peers with B and C - B peers with A - And C peers with A. 

Except for local traffic, all external and inter-office (inter-branch) traffic goes 

via Box A. – with the appropriate dial plan of course. 

Both the above methods, while useable for a basic configuration, will not provide you with 

a complete solution. To provide a complete solution is beyond the scope of this 

document. 

The following link will provide further reference for connecting two Asterisk boxes together 

http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers 

If you require a complete solution tailored to your exact requirement, my advise to you is 

to hire a VOIP consultant. 

30.5 REDIRECT INCOMING IN BOX A TO BOX B 

There will be times when you will want to redirect certain calls made to one Trixbox box to 

another Trixbox box in another location. This can happen in a scenario where you are 

moving your mail office from your existing location to another location (or moving your 

sales office to another location away from your main office). 

Personally I really wouldn’t do this as it is simpler to get a redirection to the new number from 

the exchange. 

In this example, lets say that all calls coming to your head office Trixbox PABX made on 

phone number 12345678 need to be sent to your Trixbox PABX in a different location. 

Let us call your Head Office PABX Box-A and your remote PABX Box-B and calls coming 

in to Box-A to be sent to Box-B. 

There are a number of ways that this can be achieved but most involved editing conf files 

and adding extra codes in the conf files. Unless you are quite comfortable with doing this, 

it can be quite daunting to the uninitiated. 

The example I am illustrating here does not involve editing any conf file. All can be done 

through freePBX. It is a little long winded but you don’t have to touch any of the conf files. 

Step 1 – (Do this in Box-A) 

i. Create an IAX Trunk 

You will only need the following in the Outgoing Setting and leave everything else 

blank. Let us call this trunk TransferTrunk. 

host=BoxB.com.au (your qualified DNS or the IP address) 

type=peer 


username=6000 (this number will be used later) 

ii. Create and Outbound Route 

After you have created the trunk, you will need to create an outbound route. Let us 

call this outbound route TransferRoute. 

In the dial pattern, you will only require one entry: 

6000 

The trunk sequence will be TransferTrunk only. 

iii. Create a new extension 

Next create a new extension e.g. 500. You will be using this extension to 

unconditionally forward all calls made to this extension, to extension 6000 in BoxB 

which we will create later. 

To unconditionally forward calls made to this extension to extension 6000, you need 

to connect to extension 500 using a softphone and dial *726000. Any calls to this 

extension will be unconditionally forwarded to extension 9000. 

iv. Create and inbound route 

DID number is 12345678. This is your phone number of calls come in to Box-A from, 

assuming that all calls that come on this number are to be transferred to Box-B. 

Set Destination to Core 500. 

That’s all need to be done in Box-A 

Step 2 – (Do this in Box B) 

i. Create a Trunk 

Trunk name: TransferTrunk. Leave the peer details blank. 

User context: Transfer-In 

You will only need the following in the Incoming Setting and leave everything else 

blank. 


ii. Create a new IAX extension 

Create a new IAX extension 6000 (this is the username defined in the outgoing 

setting in Step 1 above). Display name: CallTransfers. 

Edit this extension context to read from-trunk (instead of from-internal). 

iii. Direct all calls to IVR 

Ensure that incoming route direct calls to the appropriate extension. In my case, all 

calls go to IVR. 

That is all we need to do and from now on, calls that come in to Box-A on your phone 

number 12345678 will be directed to extension 500 of Box-A. However, since all calls to 

extension 500 has been unconditionally redirected to extension 6000, asterisk will forward 

the call to extension 6000 which your outbound route will pick up and send to Box-B. 

When it gets to Box-B, it will be answered by the IVR in Box-B. 


31 CUSTOMISED VOICE 

The default pre-recorded voice prompts and announcements in TRIXBOX are suitable for 

almost every situation, however there will be times when customised voice prompts or 

announcements will be required, e.g. to mix English and other language on a single 

announcement. 

Although there are separate voice sets that can be used, sometimes we only require a 

few prompts and not the entire voice set. 

31.1 CUSTOMISING VOICE PROMPTS 

To satisfy this requirement, custom voice prompts can be recorded individually by using 

the System Recording facility of freePBX as covered in the chapter referring to System 

Recording earlier in this document. 

However, if you want to manually cut the script, you can do the following (personally I 

wouldn’t bother – why re-invent the wheel). Under the [from-internal-custom] context 

of extensions_custom.conf, add the following codes. 

[from-internal-custom] 

; For custom recording 

exten => 5678,1,Wait(2) 

exten => 5678,2,Record(/tmp/my-recording:gsm) 


; for playback of custom recording 

exten => 5679,1,Playback(/tmp/my-recording) 


To start recording, use one of the phone extension and dial 5678. At the beep, start 

recording the voice prompt. 

The voice prompt will be saved as my-recording.gsm (gsm format) in the /tmp directory. 

When completed, hang up and dial 5679. The voice prompt will be played back. 

If you are not satisfied, repeat the above process. 

Once you are satisfied, rename the file to something related and recognisable e.g. 

my_office_business_hours.gsm. The file should then be moved to the 

/var/lib/asterisk/sound directory. 

The file can be played through your custom applications or prompts using the Playback 

or Background function of Asterisk. 

A good information on Asterisk sound files and how to create them can be found here 

http://voip-info.org/wiki/view/Asterisk+sound+files 


31.2 INSTALLING OTHER LANGUAGES 

If you wish to use a different language other that the default American English that comes 

with Asterisk, there are a couple of ways that you can do but as this guide is for 

beginners, I will go the easiest method. 

Create a subdirectory for the particular language (eg: ‘au’ for Australian English, ‘fr’ for 

French, ‘it’ for Italian etc.) in the /var/library/asterisk/sounds directory, e.g. 

/var/library/asterisk/sounds/au and send all your prompts sound files there. Also create 

the language directories in the digits, letters etc directories, and send the entire prompt 

files there. 

Add the following line in the iax.conf and sip.conf configuration file under [general]. In 

zapata.conf configuration file the line should be added under [channels]. 

language=au ; (or fr, it or whichever language is installed) 

 

You may use the Edit Configuration facility of TRIXBOX, otherwise you can use 

nano at the command prompt of your Asterisk box (or SSH to your Asterisk box) to 

edit the files that resides in the etc/asterisk directory. 

31.3 CONVERTING WAV FILES 

For those who wish to record their own sound prompts, the following may be of some 

assistance. 

Trixbox will be able to play virtually any sound that it has the codec for e.g. wav(pcm), 

wav49, gsm, g711, g729 etc. However, the gsm format seems to be the common format 

used for the default voice prompts. 

31.3.1 Converting WAV to GSM 

Since the WAV format is the most common format that is being adopted when 

recording with a windows based PC, and most times, the windows based WAV format 

does not play well with asterisk, below is a method of converting the windows 

recorded WAV format to GSM using SOX. 

Your result will be better if you record your sound files in mono, 16 bit, 8000 Hz. 

After recording the WAV sound files, transfer the sound files to the TEMP directory of 

the Trixbox PC. In this example, let,s call one of the sound files hello.wav. 

 

 

Login as root and change directory to the directory where you have transferred 

the sound files to e.g. cd /tmp 

At the prompt issue the following commands. 

• If your sound files were recorded in mono, 16 bit, 8000 Hz 

sox hello.wav hello.gsm 

• If your sound files were recorded in mono but NOT 16 bit, 8000 Hz 

sox hello.wav -r 8000 hello.gsm resample -ql 


• If your sound files were recorded in stereo, you will need the –cl switch. 

sox hello.wav -r 8000 –cl hello.gsm resample –ql 

• if your sound files were recorded in ADPCM wav files, to convert to standard 

wav file; 

sox hello.wav -r 8000 -c1 -s -w hello-out.wav resample -ql 

 

Next, move the sound file to the sound directory where all your prompts are 

stored and you are done. 

You may also convert a number of WAV files at once using the following command. 

In this example, lets assume that the files were all recorded in stereo; 

for a in *.wav; do sox "$a" -r 8000 -c1 "ècho $a|sed - 

e s/wav//`gsm" resample –ql 

31.3.2 Converting WAV to SLN 

Starting from Asterisk version 1.2.0, the .sln (SLINEAR) format seems to be the 

format that is being adopted (the GSM format is still commonly being used), so don’t 

worry about converting your existing prompts. 

To convert wav file to sln, use the following command: 

sox hello.wav -t raw -r 8000 -s -w -c1 hello.sln 

Further reference for converting wav sound files can be found here at voip-info.org. 


32 MOH (MUSIC ON HOLD) 

Trixbox comes with a number of preinstalled music on hold selections, however you may 

want to add your own music to the existing selection or remove the default selection 

altogether, and only use your own selection instead. 

32.1 CUSTOMISING MOH 

TRIXBOX supports native MP3. To change or add to your music on hold collections, 

simply upload all your MP3 to TRIXBOX through the Onhold Music screen of freePBX. 

You may also need to add the following, under [channels] in your zapata.conf file: 

musiconhold=default 

Pretty much any mp3 will work with Trixbox. Prior to that, it's better to convert it to a 

standard format. When converting my MP3 music, I use FreeRip and convert the music 

as per the following: 

Bit Rate: 128 

Mono 

Constant Bit Rate (CBR) 

Make sure that there is no ID3 Tagging. 

I found that this conversion gives me the most satisfactory result. 

To customise your music to your favourite, you may need to create a new music category 

directory to hold your favourite music. Do this through the freePBX “On Hold Music” GUI 

interface. I call my category favourites. 

Upload your favourite music to this directory. Again, do this through the freePBX GUI. 

Once finished uploading, click the “Enable Random Play” button and the red bar. 

Make a little change to your musiconhold.conf file like the example below (this way you 

don’t have to delete the music in your default directory). 

You will now have to get used to the newer approach (it no longer uses the [classes] tag): 

The musiconhold.conf files in 1.2 is now set out differently as per the example of my file 

below: 

musiconhold.conf 

; 

; Music on hold class definitions 

; This is using the new 1.2 config file format, and will not work with 

1.0 

; based Asterisk systems 

; 

[default] 

mode=files 

; 

; valid mode options: 

; quietmp3 -- default 

; mp3 -- loud 

; mp3nb -- unbuffered 

; quietmp3nb -- quiet unbuffered 

; custom -- run a custom application 

; files -- read files from a directory in any Asterisk supported format 

; 

directory=/var/lib/asterisk/mohmp3/favourite ;

andom=yes 

#include musiconhold_additional.conf 

The corresponding entries can be found in musiconhold_additional.conf – see below: 

musiconhold_additional.conf 

[acc_1] 

mode=files 

directory=/var/lib/asterisk/mohmp3/acc_1/ 

random=yes 

32.2 MOH FOR DIFFERENT TRUNKS 

So you are hosting three different sections of your business on your Trixbox with three 

different Phone numbers (or trunks for that matter) and would like to have different music 

on hold for the different Music On Hold for each of the section. 

While freePBX does not have a ready method of doing it from the GUI, we can trick the 

system into doing it with a simple creative modification. Here goes. 

Create 3 separate music categories (Rock, country and classical) and populate them with 

MP3s or whatever that turns you on. 

Create 3 custom context in extensions_custom.conf 

[custom-department_1] 

exten => s,1,playback(pls-wait-connect-call) 

exten => s,n,SetMusicOnHold(country) 

exten => s,n,Dial(SIP/2001,60,r) 

exten => s,n,Macro(vm,2001); voice mail to extension if no answer 

exten => s,n,Hangup 



exten => s,n,SetMusicOnHold(classical) 






exten => s,n,SetMusicOnHold(Rock) 




Assuming 2001,2002 and 2003 being the extensions of the various departments, 

Incoming route for Trunk1 send to 

Custom App custom-department_1,s,1 

Incoming route forTrunk2 send to 


Incoming route forTrunk3 send to 



32.3 STREAMING MUSIC ON HOLD 

Streaming music on hold is not directly supported using the GUI of Trixbox or freePBX, 

however if you are prepared to do a little editing of the musiconhold.conf file and 

downloading some external files, you will be able to get it working somehow. 

There are 2 ways of doing this and I will outline them both below: 

Method 1 

(old method of doing things but pretty straight forward and it works) 

Below is what I have to do to get Streaming On Hold for my system. I realised that it 

is not the best method, but for the sole purpose of getting Music on Hold, it got me 

working. 

From command line, find out if you have mpg123 installed using the following 

command. 

mpg123 –v 

You will find out if you have mpg123 installed or not and its version number. The 

version that we want is mpg123-0.59r. 

Since I do not have mpg123 installed, first I have to download and install mpg123 - it 

is not included in Trixbox implementation of Asterisk. To get mpg123 I have to log on 

as root and at the command prompt I did the following: 

cd /usr/bin 

wget http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz 

tar –zxvf mpg123-0.59r.tar.gz 

cd mpg123-0.59r 

make linux 

make install 

ln -s /usr/local/bin/mpg123 /usr/bin/mpg123 

I have to make a directory for the streaming music. You can do this though the 

freePBX GUI or manually. In this example, I have shown the manual method. 

su asterisk 

cd /var/lib/asterisk/mohmp3/ 

mkdir stream 

Next I need to create a zero length dummy MP3 file for this purpose in the 

/var/lib/asterisk/mohmp3/stream directory (this can be done manually). 

cd stream 

touch stream.mp3 

exit 

Now I have to do some editing of my musiconhold.conf file. 

Open the musiconhold.conf file using the Config edit option on your Trixbox and add 

the following lines on top of the file; 

[classes] 

default => quietmp3:/var/lib/asterisk/mohmp3/stream,http://128.177.3.80:4064/ 

and leave everything else out (or commented out) like my example below. 


The http://128.177.3.80:4064/ (SKY.FM – Country) is where I get my streaming 

music from. 

This is what my musiconhold.conf looks like. 

[classes] 

default => quietmp3:/var/lib/asterisk/mohmp3/stream,http://128.177.3.80:4064/ 

; 

The lines in BOLD are the lines that was added to the musiconhold.conf file. 

After having done that, reboot your system . 

The above is not necessarily being the best way to approach it but that is the simplest 

way that I know of. 

Method 2 

(as suggested by Bob Fryer) 

This method was tested on system running Elastix 0.9.2, but we expect that it will 

work just as well under Trixbox. 

cd /usr/src 

wget http://dag.wieers.com/rpm/packages/libmad/libmad-0.15.1b-4.el5.rf.i386.rpm 

wget http://dag.wieers.com/rpm/packages/libid3tag/libid3tag-0.15.1b-3.el5.rf.i386.rpm 

wget http://zak.googlecode.com/files/madplay-0.15.2b-1.el5.i386.rpm 

rpm -ivh libmad*.* 

rpm -ivh libid3tag*.* 

rpm -ivh madplay*.* 

cd /var/lib/asterisk/mohmp3/ 

mkdir stream 

cd stream 

touch stream.mp3 

Add the following to /etc/asterisk/musiconhold_custom.conf (if other lines are in this 

file, just add these afterwards) 

-------------------------------- 

[easylistening] 

mode=custom 

application = /usr/bin/madplay -q -s --mono -A 3 --attenuate=-3 -r 8000 -f 8192 -b 0 

http://128.177.3.80:4064/ 

[classical] 

mode=custom 


http://216.126.84.50:8000/ 

[Country] 

mode=custom 


http://64.236.34.106:80/stream/1075/ 

[Dance] 

mode=custom 


http://193.138.205.194:8014/ 

-------------------------------- 


The above are just examples, you can add more, or change the ones above. You 

may want to find some low bandwidth streams that are closer to your Country to 

eliminate any dropouts etc. 

Just in case formatting is incorrect, between the start of the word "Application" and 

the end of the HTTP address is all on one line. The same with the WGET. 

Reboot your system afterwards. 

Then go into your trunks and extensions in freepbx and choose the MOH Stream you 

want for on hold music. 

The Madplay application offers a little more control, especially in volume/attenuation. 

The following are addresses where you may find some streaming music for your 

MOH. 

181.kickin’ country http://64.236.34.106:80/stream/1075/ 

Classical 96.3 FM http://216.126.84.50:8000/ 

SMOOTHJAZZ.COM http://scfire-chi0l-2.stream.aol.com:80/stream/1005/ 

Sky FM – Modern Jazz http://205.188.215.227:8008/ 

Energy 98 – Dance Hit http://193.138.205.194:8014/ 

Life FM – Adelaide Australia http://203.16.214.174:8040/ 

Groove Mix http://63.249.3.194:8010/ 

The 80s channel http://64.236.34.106:80/stream/1040/ 


33 BACKUP AND RESTORE 

Backup and Restore are two of the mandatory functions of any application. FreePBX is 

no exception. With this in mind, the developer of freePBX has incorporated this facility in 

freePBX under the Tools section. 

Provision for flexibilities was built in this tool. You can schedule your backup, Daily, 

Weekly, Monthly, Ad-Hoc and even hourly if you wish to. Therefore there is no excuse for 

not making a regular backup. 

33.1 SCHEDULE A BACKUP 

To schedule a backup, from freePBX choose Tools -> Backup & Restore 

You will then be presented with the following screen: 

Next select Add Backup Schedule 

The screen that follows is where you define your backup requirement. 


I have chosen to schedule a Weekly (on Sundays t ) backup as per the illustration above: 

• In the Schedule Name, I have given it a name Weekly (to signify that it is a 

weekly job) 

• I have also selected all the items to be backed up – VoiceMail, System 

Recordings, System Configurations and CDR only. 

• Next from the Run Backup drop down choice, I have chosen Weekly (on 

Sundays) 

Once I have done these, I click on Submit Changes. 

Naturally you can schedule numerous backups. It is up to you to decide. If you are 

wondering where it is being stored, it is here /var/lib/asterisk/backups. The backup files 

are identified by the backup names. 

 

If this is the first time you schedule a backup, it is advisable to do the backup right 

away by selecting Now from the Run Backup drop down selection box. After this is 

done, you may start scheduling your regular backup by selecting the appropriate 

option presented to you. 

. 


33.2 TO RESTORE A BACKUP 

Backup itself is not useful if it cannot be restored. The freePBX developer has thought of 

this too. 

This is a very simple procedure. All you need to do is click on the Restore option and the 

rest are all a matter of choosing what it is that you want to restore. 


34 SUGARCRM 

SugarCRM is a contact management software that comes bundled with Trixbox. 

To set up SugarCRM, first, you need to open the SugarCRM application 

http:///crm using the default username of Admin and the 

password of password. 

For security reason you should change the Admin password. To do this, click on 'My 

Account' in the upper right-hand corner, then click on the 'Change Password' button 

underneath 'Users: Administrator (Admin) in the center-left of the screen. 

Change it to a new password and confirm your new password and click 'Save.' 

Now it's time to set up your contacts. I will start off setting up a couple of my internal 

extensions. 

Click on 'My Account' again and then click the 'Edit' button. 

Change 'Asterisk Phone Extension' to your Asterisk extension. My extension is 2001. 

While you are at it, change your time zone and date format as well. 

Click 'Save' to save that information. 

Let's add another one. 

Click on the contacts tab and then select 'Create Contact' from the left hand Shortcuts 

menu. 

Add another extension, in my case I chose my daughter’s extension 2002: 

Firstname: Norsurya 

Last name: Sharif 

Home: 2002 

Click 'Save' to save that information. 

Add another and another if you want to, using the method above. 

At this point, you may find that you are unable to make a phone call through SugarCRM. 

This is due to a little bug in the popup_picker.php (this bug may have been fixed by the 

time you read this, but at the time of writing, this bug exists). 

To fix this bug, you need to edit popup_picker.php by doing the following: 

From your Linux CLI, log in as root. 

cd /var/www/html/crm/modules/Contacts 

nano Popup_picker.php 

Browse down to line 121 and change it from: 

$number = preg_replace ( "/[^\d\*]/", "", $number ); 

To 

$number = preg_replace ( "/[^\d\*]/", "", $display_number ); 


You should now be able to dial from SugarCRM to your other internal extensions and to 

the outside world. 

 

You can add multiple users who will each have their own settings/contacts etc. 


35 USB PHONE SUPPORT 

Most USB Phones are pre-configured for use with Skype and as a result some USB 

Phone users are experiencing difficulties using the USB phones dial pads to dial out using 

providers other than Skype especially when using SIP. Special driver are usually 

required. 

The following are links, which may interest USB phone users, where appropriate drivers 

may be found. Users may have to try different drivers to find a suitable one. 

http://www.yealink.com/english/support.htm 

http://www.packetizer.com/products/ 

http://www.sednacomputer.com/products/se-p1k.html 

http://www.welltech.com/product_e_06.htm 

http://www.waytec.com.au/download.htm 

http://www.pc-telephone.com/free-phone.htm 

http://www.sjlabs.com/usbphone/SJphoneDriverATCOM-AU-100.exe 


36 CODEC (CODER DECODER) 

When making a call over the net, the software (soft-phone) or hardware needs to use a 

codec so as to send/receive information in a certain format and convert it to what you 

hear. What this means to you and me is that codecs compress data, allow you to transmit 

the compressed data which is then uncompressed at the receiving end. If you compress 

something with G729 and you don't have that installed, then you wont be able to 

decompress it and vice versa. 

 

g729 does not come preinstalled with Asterisk – you will need to purchase it. 

Different codecs have different compression ratios resulting in different bandwidth 

requirements. Generally the higher the compression the more CPU power required to 

compress, and sometimes decompress. 

In most cases this means: 

• Higher compression = More CPU power = Lower Bandwidth 

• Lower compression = Less CPU power = More Bandwidth 

So the more bandwidth you have, the less need for a higher compression ratio and the 

less powerful the CPU that will be needed. 

Most VoIP providers/hardware/licensed software will support G711 and G729 (However 

be sure to check this before purchasing hardware, or signing up with a VoIP provider!), 

with G711 requiring almost 3 times as much bandwidth, and providing excellent quality 

and the G729 providing quality similar to a good mobile call but using less bandwidth. 

Below are the approximate bandwidth requirements for Several Common VoIP 

Compression Algorithms using SIP. They vary slightly depending on the protocol used. 

Codec 

Codec Bit Rate Nominal Ethernet Approx. MBytes 

(Kbps) 

Bandwidth (Kbps) usage per hour 

G.711 64 87.2 39.24 

G.729 8 31.2 14.04 

G.723.1 6.4 21.9 9.86 

GSM 13.2 28.7 approx 12.92 approx 

iLBC 15.2 30.83 approx 13.87 approx 

G.723.1 5.3 20.8 9.36 

G.726 32 55.2 24.84 

G.726 24 47.2 21.24 

G.728 16 31.5 14.18 

Codecs used by TRIXBOX reside in the /usr/lib/asterisk/modules directory. 

The following links will give you some indications of the bandwidth usage of the various 

codecs that you may be using or contemplating on using. 

http://www.asteriskguru.com/tools/bandwidth_calculator.php 

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml 

http://www.voip-calculator.com/calculator/lipb/ 

http://www.connect802.com/voip_bandwidth.php 

http://www.newport-networks.com/pages/voip-bandwidth-calculator.html 

There are others. All you have to do is Google for them ☺ 


36.1 INSTALLING G729 AND G723 CODECS 

If you have the licence for the g729 codec and found that the digium version of g729 

is a little problematic and would like to try the free version, following is the method of 

installing it as described by David Klaverstyn – thanks David. 

DISCLAIMER: 

You might have to pay royalty fees to the G.729/723 patent holders for using their 

algorithm. 

Before downloading the codec module you need to verify what your processor type is, 

as to get a compatible compiled codec. 

From the Linux command type in: 

# cat /proc/cpuinfo 

Look specifically at the model name and the flags. The following is an example of the 

process that I am using. 

Process: 0 

vendor_id: 

GenuineIntel 

cpu family: 6 

model: 8 

model name: 

Pentium III (Coppermine) 

stepping: 3 

cpu MHz: 648.110 

cache size: 

256 KB 

fdiv_bug: 

no 

hlt_bug: 

no 

f00f_bug: 

no 

coma_bug: 

no 

fpu: 

yes 

fpu_exception: yes 

cpuid level: 2 

wp: 

yes 

flags: 

fpu vme de pse tsc msr pae mce cx8 mtrr pge mca 

cmov pat pse36 mmx fxsr sse 

bogomips: 1297.41 

From this information we can see that it is a Pentium III processor and it is using sse. 

This is the 2 bits of required information that we need. If you don’t see sse in your 

flag then the file you need must contain the words no-sse. 

Now go to http://kvin.lv/pub/Linux/Asterisk/ and select wither Asterisk version 1.2 or 1.4 

depending on the version you are using. 

The files are named as codec_g72x--pentiumx.so 

So from the information above to obtain the G729 codec we need to get the file 

codec_g729-gcc-pentium3.so and for the G723 codec we need to get the file g723- 

gcc-pentium3.so 

The gcc represents the way the module was compiled. This is your most likely 

choice. To view the different type of compiled version look at this file 

http://kvin.lv/pub/Linux/Asterisk/README 

Now that we know what file we need you can now download the appropriate file. 

Change folders to /usr/lib/asterisk/modules 

# cd /usr/lib/asterisk/modules 


Download the required file(s) 

For the G729 codec 

For Pentium III 

# wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-ast14-gcc4- 

glibc-pentium3.so 

For Pentium IV 



For the G723 codec 

For Pentium III 



For Pentium IV 

# http://kvin.lv/pub/Linux/Asterisk/ codec_g723-ast14-gcc4-glibcpentium4.so 

Rename those files to codec_g729.so or codec_g723.so accordingly. 

Restart asterisk and you are done. 

If Asterisk does not start then you may need to try icc files instead of gcc. 

36.2 TO INSTALL CODECS ON PIAF 

All processors EXPERIMENTAL and please ensure that this script is only used on 

PiaF or the codecs may not work. (unfortunately celeron processors are not supported). 

• Log onto PBX in a Flash via the CLI (Command Line Interface) as user root 

cd /root 

• wget http://www.pbxinaflash.org/scripts/install-codecs.x.gz 

• gunzip install-codecs.x.gz 

• chmod +x install-codecs.x 

• ./install-codecs.x 

• In the program you will have to agree to the license before the install can 

continue 

• you will then be prompted to choose the type of processor you have. You 

must choose the correct processor or else Asterisk will stop working! 

• rm -f install-codecs.x 

To check if the codecs has been properly installed, at the Asterisk CLI type the 

following: 

core show translation 



37 PSTN TRUNK – SPECIAL HANDLING 

37.1 DIAL 0 FOR OUTSIDE LINE 

There will be times when a user would like to get an outside line something similar to a 

PABX by simply dialling a number e.g. 0 or 9. 

I got around it by creating an entry in extensions_custom.conf. 

In my case I use 0 as the prefix to get an outside line. 

I have also pre-recorded a voice prompt to instruct the caller what to do after the second 

dial tone. 

37.1.1 Using Zaptel Card Method 

;############################################## 

; To get outside line through ZAP - Ben 22-12-2005 

; after dialling 0 will get a second dial tone. 

; Then dial the number required. 

; No need to press the send button after the second number dialled 

;############################################## 

exten => 0,1,Playback(start-dial-after-dialtone) 

exten => 0,2,Dial(ZAP/1/{EXTEN}) 


The example above is based on systems that have Zaptel card (e.g. TDM400 or 

X100P etc.) installed. Note:* If you're using a PRI line, you need to enable 'overlapdial' 

in /etc/asterisk/zapata.conf for this to work. 

37.1.2 Using SPA3K Method 

Using SPA3K will be a little tricky. I am describing how I got around it (assuming that 

you have set up the SPA3K as described in this document), however there maybe 

other method. 

1. In the SPA Line 1 Tab, I have included the following entry in the dialplan: 

) 

2. The following is what my dialplan looks like in SPA3K 

(**|*xx|**x.|xx.|) 

3. In the PSTN Line Tab, I have the following: 

• VOIP-PSTN Gateway Enabled: Yes 

• VOIP Caller Auth Method: None 

• VOIP Caller Default DP: None 

In made the following entry in my Asterisk entensions_custom.conf 

;############################################## 

; To get outside line through SPA3K - Ben 25-12-2005 

; after dialling 0 will get a second dial tone. Then dial the 

; number required. No need to press the send button again 

;############################################## 


exten => 0,1,Playback(start-dial-after-dialtone) 

exten => 0,2,Dial(SIP/Telstra/{EXTEN}) ; Telstra being the name of the SPA3K trunk. 


Every time when I dial ‘0’, the SPA3K will send it to Gateway 0, which is the PSTN 

line and present me with a second dial tone. 

37.2 GIVING PRIORITY TO EMERGENCY NUMBERS (E.G. 000) 

Emergencies, by definition should be given right of way. This applies to the right of using 

the telephone for emergency purposes above other uses. 

Imagine what will happen if you want to use the PSTN line to call emergency 000 (in 

Australia) and someone else was hogging the line. The sensible way is for Asterisk to 

disconnect the line and allow you to make the emergency call instead. 

The following method will provide you with the facility, however you need to do some 

editing to the extensions_custom.conf file in /etc/asterisk. 

In my case I have pre-recorded a voice prompt called ‘calling-emergency’ which says, 

“You are trying to call emergency assistance. If you have dial this in error, please hangup” 

;################################################### 

; Give priority to emergency number by disconnecting existing user 

; if using the PSTN line first, otherwise simply call the emergency number. 

;################################################### 

exten => 000,1,Background(calling-emergency) ; voice warning 

exten => 000,2,Wait(1) ; give caller a chance to hang up 

exten => 000,3,Dial(Zap/1/000,,j) ; If ZAP 1 is busy, jump to n+101 

exten => 000,4,Hangup() ; hang up after the call 

exten => 000,104,Softhangup(Zap/1-1) ; hang up the line if being used 

exten => 000,105,Goto(3) ; make call after disconnecting the other user 


38 PORT FORWARDING AND QOS – ROUTERS 

38.1 PORT FORWARDING 

While most domestic routers using UPnP do not require port forwarding, some routers do. 

Even if your router does not require port forwarding for TRIXBOX to work, do yourself a 

favour and forward a few of the relevant ports to your Asterisk IP address anyway – it 

doesn’t hurt. 

IAX Port: 4569 (UDP) 

Sip ports: 5004 - 5082 (UDP and TCP), more if you want to. 

RTP ports: 10000 – 10500 (UDP) – this should be enough. 

I am using the LinkSys WRT54G wireless router. In the example below, I have forwarded 

all the ports mentioned above to my asterisk box IP, which is 192.168.1.101 

These should be enough for starters. 

38.2 QOS – ROUTERS 

Some routers have QOS facility while some routers haven’t. It is highly recommended 

that you enable this function if your router supports it. 

There have been a number of discussions on the effectiveness of QOS on broadband 

utilising the public Internet circuit. It is beyond this guide to document the effectiveness or 

otherwise of enabling QOS on your router. For a start, there are just too many routers 

there for me to give any meaningful examples. 

I use the Linksys WRT54G®. Taking the easy way out, I simply restrict the PCs (see 

below) that more often than not will be used as a P2P client or chatting on the MSN chat 

channels. 

I restrict the PCs according to their MAC addresses. 


In the application priority section (see below), I have defined the various ports that I want 

priority to be accorded the highest priority. In this case I have defined the port ranges 

related to my VOIP requirement. 

My recommendation is, if your router has the function, use it. At the very least, it will 

prioritise traffic within your own LAN especially when you have sons and daughters (for 

that matter you too can be the culprit) who are always on the P2P, leeching files and 

music. 

 

This may not be the best solution. If anyone has a better solution for the Linksys 

WRT54G, please let me know. 


39 UPGRADING YOUR SYSTEM 

Mostly everyone would like to have the latest and the greatest and if you are like most, 

chances are that you would certainly want to update your TRIXBOX and/or freePBX as 

soon as there is a new update. 

If you are currently using Trixbox v2.x, the following instruction may get you there. It’s 

best to stop asterisk. Through SSH or the command prompt: type: 

amportal stop (Remember to always stop asterisk first before doing this) 

service zaptel stop (it helps to stop the zaptel service as well) 

39.1 UPGRADING TRIXBOX 

The trixbox-update.sh command has been depreciated and no longer availavle. To 

upgrade Trixbox, you now simply do it through yum. 

yum –y update 

At the end of it, you may need to reboot. 

reboot. 

39.2 UPGRADING FREEPBX 

There are 2 ways to upgrade freePBX as explained at the aussievoip website here 

http://www.aussievoip.com/wiki/freePBX 

Either one will be fine but I would like to suggest that you take the all-in-one version with 

modules and be done with it. Afterall, this is the method recommended if one is on a 

limited-bandwidth connection, or the FreePBX machine doesn't have easy Internet 

access. 

To conduct this upgrade, you will need to log on to the Trixbox PC as root. 

At the command prompt, do the following; 

cd /usr/src/ 

wget http://mirror.freepbx.org/freepbx-2.4.0.tar.gz 

tar zxvf freepbx-2.4.0.tar.gz 

cd freepbx-2.4.0 

./install_amp 

Once this is done, freePBX is now upgraded to version 2.4.0 and you can start managing 

and/or upgrading the freePBX modules. 

The other method is through the upgrade mudule that can be obtained through Module 

Admin of freePBX. This is an easier method for most. 


40 TOOLS 

The following are tools that I use regularly to maintain my Asterisk systems. 

40.1 WEBMIN FOR TRIXBOX 

Webmin is an invaluable web based tool for managing a Linux box. Webmin make it easy 

to configure application like SMTP mail, editing files etc. 

Webmin is not installed as default in Trixbox but it is in the repositury that user can 

download. However I found that the package manager can be extremely slow if not 

impossible to get into. As a result I have resorted to doing it manually and much faster 

too. 

At the command line simply issue the following command: 

yum install webmin 

I have found the above method is straightforward and simple. However there are some 

users who found that following an alternative method is simpler. If that is the case, the 

alternative installation method can be found here: 

http://www.terrasoftsolutions.com/support/solutions/ydl_general/webmin.shtml 

Once installed, you may connect to Webmin remotely through your browser using the 

following address https://:10000. E.g. 

https://192.168.0.101:10000 

If you would like to avoid clashing the port with your RTP port which is 10000 – 20000, 

you may edit the file /etc/webmin/miniserv.conf and change the port value to something 

else, such as 9000. 

Locate the entry where it says: 

Port=10000 

And change this to; 

Port=9000 

You can use https://192.168.0.101:9000 to connect to Webmin instead. 

The alternative is to edit the /etc/asterisk/rtp.conf file like the example below and you 

can keep https://192.168.0.101:10000 


This will break PA1688 based phones. It's better to change your webmin.conf file 

[general] 

; 

; RTP start and RTP end configure start and end addresses 

; 

rtpstart=10000 < - - change this to 10002(totally unnecessary.. ☺) 

rtpend=20000 

40.1.1 To update WebMin 

Anytime you want to update Webmin, simply do the following. 

Log on to your Asterisk box (SSH or at the console). 

At the command prompt, issue the following command: 

yum –y install webmin 

40.3 PUTTY 

PuTTY is a free implementation of Telnet and SSH for Win32 and Unix platforms, along 

with an xterm terminal emulator. 

It is written and maintained primarily by Simon Tatham and can be downloaded from the 

following link.http://www.putty.nl/download.html 

40.4 WINSCP 

WinSCP is an open source freeware SFTP client for Windows using SSH. Legacy SCP 

protocol is also supported. Its main function is safe copying of files between a local and a 

remote computer. 


It can be downloaded from the following link. 

http://winscp.net/eng/index.php 


41 USE ACCESS TO CONNECT TO DATABASE 

A quick and dirty way to access the Trixbox database is through MSACCESS. This will 

enable you to work on the data kept in the MySQL database which Trixbox uses. You 

may want to extract CDR data for reporting etc. Imagination is the restricting factor. 

To do this first you must download the MySQL ODBC driver for Windows here: 

http://www.mysql.org/downloads/connector/odbc/3.51.html save it somewhere in your 

Windows local hard drive but do not install it yet. 

41.1 SETTING UP A NEW MYSQL USER 

Log into the TrixBox web interface, and navigate to the phpMyAdmin screen. 

phpadmin selection on Trixbox 

When selected, you will be presented the following screen. 

Select and Click "Privileges" (see arrow in the illustration above) and a new screen will 

pop up – see below. 


Select and Click "Add a new user" (see illustration above) and you will presented with the 

following screen but all the fields will be blank. The arrows indicate all the fields that I 

have filled. 

In the blank fields enter the following: 

User name: enter a User name (e.g. reportis) 

Password: enter a password of your choice (e.g. passw0rd) 

Since the database lives in the same host, leane the Host field blank. 

Under the Data section of "Global Privileges" – see arrow, check all the options. Since we 

don’t really want the user to start playing oround with the structure and other aspects of 

the database, leave the Structure and Administrations unticked. 

Once done, click "Go" in the lower right corner of the screen to save the new user after 

which you can get out of PHPAdmin. 

41.2 CONFIGURE MYSQL ODBC FOR WINDOWS 

Install the ODBC Driver that was downloaded. 

Once installed, on the Windows PC, click Start -> Control Panel -> Performance & 

Maintenance -> Administrative Tools -> Data Sources (ODBC) 

The following screen will appear. 


Select the Uer DSN tab and click on the “Add” button. 

Scroll to the bottom of the list, and select "MySQL ODBC 3.51 Driver" (see arrow), then 

click the Finished button. 

When following dialog box appears, fill in the following information: 


Data Source Name: TrixBox CDR 

Description: Call Detail Reporting 

Server: 192.168.1.100 (your server IP address) 

User: reports (the username you created in phpMyAdmin previously) 

Password: passw0rd (the password that we entered in phpadmin earlier) 

Database: asteriskcdrdb 

Once this is done, click "Test" to ensure that the connection is working 

Click "Ok" to save and finish and “OK” again to close the ODBC dialog box. 

41.3 CONNECTING USING MSACCESS 

To connect to the CDR database using MSACCESS, you must first create a blank 

database. Once the blank database has been created, the CDR table from MySQL can 

then be linked to the MSACCESS database that you have created. Once doce, you can 

start working on the data. Those familiar with Access can start creating a special 

application for their use. 

Create an empty ACCESS database. Call it reporting or whatever you like. 


Select Tables as illustrated by the arrow above. 

Select and click on File (see red arrow). The next screen will appear. 


Select “Get External Data” and choose “Link Tables…” and click. 

On the explorer’s screen that follows, click the drop down on the “Files of type” and select 

“ODBC Database()” – see illustration below. 

On the screen that follows, select the “Machine Data Source” tab and select Trixbox CDR 

and click on “OK”. 

When the following dialog box appears, select “cdr” and click “OK” 

Click “OK” again when the following dialog box appears. 


When you press OK, the CDR table will be linked to the blank Access Database and you 

can start looking at the data by clicking on the “cdr” table. 

Now if you click on the “cdr” table and it will open and display the data in the table in 

MSACCESS – see below. 

View of CDR table through MSACCESS 


IBM 

TALK / DATA 

TALK 

RS CS TR RD TD CD 

1 2 3 

4 5 6 

7 8 9 

* 8 # 

1 2 3 

4 5 6 

7 8 9 

* 8 # 

1 2 3 

4 5 6 

7 8 9 

* 8 # 

42 MY TRIXBOX® PBX NETWORK 

The following is a representation of my network schematic on my Local Area Network with 

my Asterisk box as I have done according to my narration above. 

My VOIP Network 

VOIP Service Provider 

PSTN 

Internet 

SPA3000 

8 Ports router 

192.168.1.1 

192.168.1.10 

Broadband Modem 

Asterisk PC (IPBX) 

192.168.1.101 

C3010-2E 

192.168.1.11 

PC 

192.168.1.21 

PC using SoftPhone 

(X-Ten) 

192.168.1.20 

C3010-2E 

192.168.1.12 

C3010-2E 

192.168.1.13 

PC 

192.168.1.22 

Through PSTN if network down 

PSTN Traffic to/from PBX 

Network Server 

192.168.1.100 

PC 

192.168.1.23 

VOIP Traffic 

ALL IP Phones and ATA registered to Asterisk 

Asterisk registered to VOIP Service Provider /s 


43 TROUBLE SHOOTING 

With any installation, you will find some unexpected problems. Some are inherent issues 

due to bugs or undocumented features, while others are unintentional mistakes. 

We will endeavour to list troubleshooting tips as we experience them (or those suggested 

by other users). These will be added progressively as we encounter them. 

43.1 DEBUG MESSAGES AND LOG FILES ARE YOUR FRIENDS 

Whenever problem strikes, there is always a reason for it. To assist you in locating the 

problem, generally there are 2 places that you can have a look at – the Asterisk Log 

Files, under TOOLS (which will give you the last 2000 entried in the log). 

The other is to monitor the activities of your Trixbox using the following command at the 

command line (which will give you the Asterisk CLI). 

Asterisk –rvvvv (the “v” depends on the level of verbosity you need) 

Studying the activities will give you an indication of what has gone wrong. 


43.2 UNABLE TO LOG ON TO PACKAGES 

From time to time, you may be having problem logging on to “Packages” to 

install/uninstall or update Trixbox modules. This has happened to me a number of time 

which made me post in the Trixbox forum thinking that I may have done something 

wrong. 

One such exampple is when I need to get the package that allow me to get tp PHPADMIN 

through thr GUI. After a number of tries I resorted to installig it manually uing the 

following command at the command line: 

yum install tbm-phpmyadmin 

This difficulty is due to the delay in connecting to the site (depending on how busy the site 

is at the time) and you get connection time out. Trying again later may solve this problem 

or you may use a hammer and fix it by changing an entry in the etc/php.ini file. 

Change the max_execution_time from 30 to 60 may solve this problem. All it does is to 

wait 60 seconds before reporting a time out. 

You may additionally delete guidRegistered using the following command: 

rm /var/www/html/maint/cache/guidRegistered 

Reboot and you should be on your way. 

43.3 ASTERISK FEATURE CODES NOT WORKING 

If you are using ATAs such as the Sipura SPA3K, Linksys PAP2 etc., you will need to 

delete all the Vertical Service Feature codes in the ATA, otherwise the Asterisk feature 

codes such as *77, *99 etc may not work. This is caused by the ATA intercepting the 

codes thinking that it is its own feature codes. 

This also applies to all other ATAs, FXS devices such as the various routers with built-in 

FXS ports and IP Phones that has their own feature codes built-in. Unless these feature 

codes are disabled, they are likely to clash with Asterisk's feature codes and as such 

Asterisk feature codes will not give you the desired results. 

43.4 ASTERISK DROPS CALLS AFTER A FEW SECONDS 

If Asterisk rings an extension but drops the calls too soon before it was answered, 

chances are, you may have set the ‘Number of seconds to ring phones before sending callers 

to voicemail:’ too short. This setting can be found in the General Setting under 

Voicemail. 

43.5 PENNYTEL ASTERISK PROBLEM 

Pennytel, for some reason rejects user agent if it is Asterisk. To work around this, you 

are required to introduce the following line in your sip.conf under [general] 

useragent = My PABX ; required if you are with Pennytel 

The user agent can be any string as long as it is not Asterisk. 

43.6 UNABLE TO RECEIVE INCOMING CALLS 

If you are unable to receive incoming calls, chances are that you may have missed setting 

up the incoming call handling as suggested in the previous sections of this guide. 


A quick fix for TRIXBOX is to create an Inbound Route with the DID and CID field left 

blank. At least this will let incoming calls through while you try to solve the problem. 

Please refer to the following section on Calls Handling in this guide: 

43.7 SIPURA SPA3000 – NOT HANGING UP AFTER PSTN CALLS 

For months I have to wrestle with the problem of SPA3K not hanging up after a PSTN 

call. Although the caller has hung up, SPA3K will hold the line off hook until I manually 

power cycle SPA3K. 

The following settings may overcome the problem: 

. 

PSTN Line Tab (Advanced Mode) 

PSTN Disconnect Detection 

Detect CPC: yes 

Detect Polarity Reversal: yes 

Detect PSTN Long Silence: yes 

Detect VoIP Long Silence: yes 

PSTN Long Silence Duration: 180 

VoIP Long Silence Duration: 180 

PSTN Silence Threshold: medium 

Min CPC Duration: 0.2 

Detect Disconnect Tone: no 

Disconnect Tone: Leave blank 

 

I deleted my original Disconnect Tone setting of: 425@-30,425@- 

30;1(.375/.375/1+2) which seems to work well for Telstra, and replaced it with 

425@-30,425@-30;1(.390/.390/1+2) which works better with Optus. 

Only one catch with this: if you put somebody on hold for longer than the above settings, 

if they don't make any noise, it will hang up. 

43.8 SYSTEM NOT WORKING AFTER CHANGING TO STATIC IP 

If you are using Netconfig to change to Static IP and to a new IP address and the system 

stopped working after that, netconfig in Centos V5 is the culprit. 

Using Nano, verify that /etc/sysconfig/network-scripts/ifcfg-eth0 contains the following. 

NAME="" 

GATEWAY=192.168.1.1 (or whatever your gateway should be) 

BOOTPROTO=static 

TYPE=Ethernet 

DEVICE=eth0 


MTU="" 

NETMASK=255.255.255.0 (or whatever your network mask is) 

NOZEROCONF=yes 

BROADCAST=192.168.1.255 

IPADDR=192.168.1.100 (or whatever your IP address is) 

NETWORK=192.168.1.0 

ONBOOT=yes 

Also check /etc/resolv.conf amd make sure you have the following. 

nameserver your primary dns 

nameserver your secondary dns 

The easiest out is to change your network setting using the Trixbox System Tab GUI or 

those who have Webmin installed, use Webmin to do it. 

43.9 NO AUDIO ON ANALOG LINE 

Problem: Call on an anlogue line seems to ring and connect correctly but no audio 

Solution: Trixbox now uses the oslec echo canceler which has provided significant 

improvement on echo problems on even the "worst" X100P clone card. In order for this to 

operate correctly echotraining needs to be commented out in zapata.conf. The installation 

does not always comment it out so it needs to be checked. The same parameter needs 

done for every FXS extension, but this is done in freebpbx GUI. After defining the zap 

extension go back and change the echotraining parameter from its default 800 to "blank". 

43.10 UNEXPLAINED CRASHES 

Problem: Unexplained crashes every few hours. People with other hardware platforms 

(Dell, Asus) also report the problem. This issue is not universal so seems to be hardware 

specific. 

Solution: Delete acpi=off in the kernel line in grub.conf. 

43.11 ZAP INBOUND ROUTE DOES NOT WORK 

Problem: Inbound route definition for a zap channel does not work, but a catch all 

inbound route definition captures the call correctly. 

Solution: This is because genzaptel.conf does not seem to insert the from-zaptel context 

in zapata-channels.conf. Check zapata-channels.conf and make sure context=from-zaptel 

is in the FXO port definition. Note, every time genzaptelconf is rerun you have to change 

it back again. 

43.12 OCCASSIONAL FATAL ERROR WHEN DEFINING EXTENSION 

Problem: Sometimes defining an extension generates FATAL ERROR. Error: Trying to 

write null voicemail file! I refuse to continue! 

Solution: Define the extension with voicemail enabled, and apply the config. Then go back 

and disable it. Alternatively copy vm_email.inc and vm_general.inc from /usr/src/freepbx- 

2.3.1/amp_conf/astetc to /etc/asterisk 


43.13 ALL EXTENSIONS FAILED TO WORK WHEN INTERNET DOWN 

One of the work around to avoid this from happening 

1. Install cache only name server on the trixbox. 

• Activate the CENTOS base packages and CENTOS base extras in the 

repositories dropdown menu under the settings tab in the dashboard 

• Load the package manager and find the caching-nameserver module 

• Tick the box to install caching-nameserver and go to bottom /press install 

2. Open an ssh window and type the following: 

cd 

chkconfig --add named 

chkconfig --level 345 named on 

echo "* IN A 127.0.0.2" >> /var/named/localdomain.zone 

service named start 

3. back to dashboard: system tab, dropdown menu NETWORKING: 

Click on OPEN NETWORK PARAMETERS 

Click on USE OPEN DNS 

Change the primary dns to: 127.0.0.1 

Make sure host is: trixbox1.localdomain 

Ensure that HOSTNAME=trixbox1.localdomain and NOT 

5. Reboot 

In this way, there is nothing you need to do when internet goes down. 


APPENDIX A 

A.1 ASTERISK FEATURE CODES 

Asterisk feature codes are defined in the feature codes selection of the Setup Tab, 

illustrated below: 

# Directory 

** Call Pickup 

*11 User Log on 

*12 User Log off 

*30 Blacklist a number 

*31 Remove a number from the blacklist 

*32 Blacklist the last caller 


*43 Echo Test 

*52 Call Forward No Answer/Unavailable Activate 

*53 Call Forward No Answer/Unavailable Deactivate 

*60 Time 

*61 Weather 

*62 Schedule wakeup call 

*65 Speak your Extension number (your extension is XXX) 

*69 Last Caller number/ID 

*70 Activate Call Waiting (deactivated by default) 

*71 Deactivate Call Waiting 

*72 Call Forwarding System 

*73 Disable Call Forwarding 

*74 Call Forward all prompting deactivate 

*77 IVR Recording 

*78 Enable Do-Not-Disturb 

*79 Disable Do-Not-Disturb 

*90 Call Forward on Busy 

*91 Disable Call Forward on Busy 

*92 Call Forward Busy Prompting deactivate 

*97 Message Center (does no ask for extension) 

*97 + 0 Change email greeting message. 

*98 Enter Message Center 

*99 Playback IVR Recording 

*411 Directory (while in IVR – similar to #) 

*422 Connect to Gabcast 

70 Call Parking (or 700 see Call Parking and Transfer) 

555 ChanSpy 

666 Test Fax 

888 Zap Barge (there are variations) 

7777 Simulate incoming call 

A.2 ASTERISK CLI COMMANDS 

These are some of the available CLI commands that can be executed from the console 

when you run: asterisk –r (or –rvvv depending on the level of verbosity you may want). 

To stop TRIXBOX – amportal stop 

To start TRIXBOX – amportal start 

A.2.1 General commands 

!: Executes a given shell command 

abort halt: Cancel a running halt 

add extension: Add new extension into context 

add ignorepat: Add new ignore pattern 

add indication: Add the given indication to the country 

amportal start: Stop TRIXBOX and 

amportal stop: Restart TRIXBOX. 

debug channel: Enable debugging on a channel 

dont include: Remove a specified include from context 

help: Display help list, or specific help on a command 

include context: Include context in other context 

load: Load a dynamic module by name 

logger reload: Reopen log files. Use after rotating the log files. 

no debug channel: Disable debugging on a channel 

pri debug span: Enables PRI debugging on a span 

pri intense debug span: Enables REALLY INTENSE PRI debugging 

pri no debug span: Disables PRI debugging on a span 


emove extension: Remove a specified extension 

remove ignorepat: Remove ignore pattern from context 

remove indication: Remove the given indication from the country 

save dialplan: Overwrites your current extensions.conf file with an exported 

version based on the current state of the dialplan. A backup copy of your old 

extensions.conf is not saved. The initial values of global variables defined in 

the [globals] category retain their previous initial values; the current 

values of global variables are not written into the new extensions.conf. 

Using "save dialplan" will result in losing any comments in your current 

extensions.conf. 

set verbose: Set level of verboseness 

show agents: Show status of agents 

show applications: Shows registered applications 

show application: Describe a specific application 

show channel: Display information on a specific channel 

show channels: Display information on channels 

show codecs: Display information on codecs 

show conferences: Show status of conferences 

show dialplan: Show dialplan 

show image formats: Displays image formats 

show indications: Show a list of all country/indications 

show locals: Show status of local channels 

show manager command: Show manager commands 

show manager connect: Show connected manager users 

show parkedcalls: Lists parked calls 

show queues: Show status of queues 

show switches: Show alternative switches 

show translation: Display translation matrix 

show voicemail users: List defined voicemail boxes 

show voicemail zones: List zone message formats 

soft hangup: Request a hangup on a given channel 

A.2.2 AGI Commands 

show agi: Show AGI commands or specific help 

dump agihtml: Dumps a list of agi command in html format 

A.2.3 Database Handling 

database del: Removes database key/value 

database deltree: Removes database keytree/values 

database get: Gets database value 

database put: Adds/updates database value 

database show: Shows database contents 

A.2.4 IAX Channel Commands 

iax2 debug: Enable IAX debugging 

iax2 no debug: Disable IAX debugging 

iax2 set jitter: Sets IAX jitter buffer 

iax2 show cache: Display IAX cached dialplan 

iax2 show channels: Show active IAX channels 

iax2 show peers: Show defined IAX peers 

iax2 show registry: Show IAX registration status 

iax2 show stats: Display IAX statistics 

iax2 show users: Show defined IAX users 

iax2 trunk debug: Request IAX trunk debug 

iax debug: Enable IAX debugging 

iax no debug: Disable IAX debugging 


iax set jitter: Sets IAX jitter buffer 

iax show cache: Display IAX cached dialplan 

iax show channels: Show active IAX channels 

iax show peers: Show defined IAX peers 

iax show registry: Show IAX registration status 

iax show stats: Display IAX statistics 

iax show users: Show defined IAX users 

init keys: Initialize RSA key passcodes 

show keys: Displays RSA key information 

A.2.5 SIP Channel commands 

sip debug: Enable SIP debugging 

sip no debug: Disable SIP debugging 

sip reload: Reload sip.conf (added after 0.7.1 on 2004-01-23) 

sip show channels: Show active SIP channels 

sip show channel: Show detailed SIP channel info 

sip show inuse: List all inuse/limit 

sip show peers: Show defined SIP peers (register clients) 

sip show registry: Show SIP registration status (when Asterisk registers as a 

client to a SIP Proxy) 

sip show users: Show defined SIP users 

A.2.6 Server management 

restart gracefully: Restart Asterisk gracefully 

restart now: Restart Asterisk immediately 

restart when convenient: Restart Asterisk at empty call volume 

reload: Reload configuration 

stop gracefully: Gracefully shut down Asterisk 

stop now: Shut down Asterisk immediately 

stop when convenient: Shut down Asterisk at empty call volume 

extensions reload: Reload extensions ONLY 

unload: Unload a dynamic module by name 

show modules: List modules and info about them 

show uptime: Show uptime information 

show version: Display Asterisk version info 

A.3 ASTERISK SPECIAL EXTENSIONS 

The following are some Asterisk Special Extensions that are useful to know when creating 

your Dial Plan or Special Custom Context. 

i = invalid, jump here when an invalid number is dialled. 

t = timeout, jump here when we timeout. 

h = hang-up, jump here when a call in hung up. 

s = standard, jump here if the dialled extension is unknown (used for generic 

processing). 

o = operator, a special extension to escape voice mail (configurable) 


A.4 ASTERISK COMMON VARIABLES 

The following are some of the special variables in use with the Asterisk scripts as defined 

in the Asterisk’s document. These variables are automatically defined when a new call 

starts. 

${ACCOUNTCODE} 

${CALLERID} 

${CALLERIDNAME} 

${CALLERIDNUM} 

${CALLINGPRES} 

${CHANNEL} 

${CONTEXT} 

${DATETIME} 

${DNID} 

${ENUM} 

${EPOCH} 

${EXTEN} 

${ENV(VAR)} 

${HANGUPCAUSE} 

${INVALID_EXTEN} 

${LANGUAGE} 

${LEN(VAR)} 

${MEETMESECS} 

${PRIORITY} 

${RDNIS} 

${TIMESTAMP} 

${TXTCIDNAME} 

${UNIQUEID} 

${SIPCALLID} 

${SIPDOMAIN} 

${SIPUSERAGENT} 

Account code (if specified) 

Caller ID 

Caller ID Name only 

Caller ID Number only 

PRI Caller ID presentation for incoming calls 

Current channel name 

Current context 

Current date time in the format: YYYY-MM-DD_HH:MM:SS 

Dialed Number Identifier 

Result of application EnumLookup 

Current unix style epoch 

Current extension 

Environmental variable VAR 

Asterisk hangup cause 

The invalid called extension (used in the "i" extension) 

Current language 

String length of VAR (integer) 

Number of seconds a user participated in a MeetMe 

Current priority 

Redirected Dial Number ID Service 

Current date time in the format: YYYYMMDD-HHMMSS 

Result of application TXTCIDName 

Current call unique identifier 

SIP Call-ID: header verbatim (for logging or CDR matching) 

SIP destination domain of an inbound call (if appropriate) 

SIP user agent 

The following are set by the dial() application: 

${DIALEDPEERNAME} Dialed peer name 

${DIALEDPEERNUMBER} Dialed peer number 

${DIALEDTIME} Total time for the call in seconds (Network time). 

${ANSWEREDTIME} Time from answer to end of call in seconds. 

${DIALSTATUS} Status of the call, one of: | CHANUNAVAIL |CONGESTION | BUSY | 

NOANSWER | ANSWER | CANCEL 


A.5 INDICATIONS FOR VARIOUS COUNTRIES 

The following are some of the indications for the various countries that I am aware off. 

Please submit others that you know of to me. 

[at] 

ringcadence = 1000,5000 

dial = 420 

busy = 420/400,0/400 

ring = 420/1000,0/5000 

congestion = 420/200,0/200 

callwaiting = 420/40,0/1960 

dialrecall = 420 

record = 1400/80,0/14920 

info = 950/330,1450/330,1850/330,0/1000 

stutter = 380+420 

[au] 

ringcadence = 400,200,400,2000 

dial = 413+438 

busy = 425/375,0/375 

ring = 413+438/400,0/200,413+438/400,0/2000 

congestion = 425/375,0/375,420/375,0/375 

callwaiting = 425/200,0/200,425/200,0/4400 

dialrecall = 413+438 

record = !425/1000,!0/15000,425/360,0/15000 

info = 425/2500,0/500 

std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100 

facility = 425 

stutter = 413+438/100,0/40 

ringmobile = 400+450/400,0/200,400+450/400,0/2000 

[br] 


dial = 425 

busy = 425/250,0/250 

ring = 425/1000,0/4000 

congestion = 425/250,0/250,425/750,0/250 

callwaiting = 425/50,0/1000 


record = 425/250,0/250 

info = 950/330,1400/330,1800/330 

stutter = 350+440 

[be] 


dial = 425 

busy = 425/500,0/500 

ring = 425/1000,0/3000 

congestion = 425/167,0/167 

callwaiting = 1400/175,0/175,1400/175,0/3500 

dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 

record = 1400/500,0/15000 

info = 900/330,1400/330,1800/330,0/1000 

stutter = 425/1000,0/250 


[ch] 


dial = 425 

busy = 425/500,0/500 

ring = 425/1000,0/4000 

congestion = 425/200,0/200 

callwaiting = 425/200,0/200,425/200,0/4000 

dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 

record = 1400/80,0/15000 

info = 950/330,1400/330,1800/330,0/1000 

stutter = 425+340/1100,0/1100 

[cl] 


dial = 400 

busy = 400/500,0/500 

ring = 400/1000,0/3000 

congestion = 400/200,0/200 

callwaiting = 400/250,0/8750 

dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400 

record = 1400/500,0/15000 

info = 950/333,1400/333,1800/333,0/1000 

stutter = 

!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/10 

0,!0/100,400 

[cn] 


dial = 450 

busy = 450/350,0/350 

ring = 450/1000,0/4000 

congestion = 450/700,0/700 

callwaiting = 450/400,0/4000 

dialrecall = 450 

record = 950/400,0/10000 

info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400 

stutter = 450+425 

[cz] 


dial = 425/330,0/330,425/660,0/660 

busy = 425/330,0/330 

ring = 425/1000,0/4000 

congestion = 425/165,0/165 

callwaiting = 425/330,0/9000 

dialrecall = 

!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660 

record = 1400/500,0/14000 

info = 950/330,0/30,1400/330,0/30,1800/330,0/1000 

stutter = 425/450,0/50 

[de] 


dial = 425 


usy = 425/480,0/480 

ring = 425/1000,0/4000 

congestion = 425/240,0/240 

callwaiting = 

!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/ 

200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0 

dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 

record = 1400/80,0/15000 

info = 950/330,1400/330,1800/330,0/1000 

stutter = 425+400 

[dk] 


dial = 425 

busy = 425/500,0/500 

ring = 425/1000,0/4000 

congestion = 425/200,0/200 

callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0 

dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 

record = 1400/80,0/15000 

info = 950/330,1400/330,1800/330,0/1000 

stutter = 425/450,0/50 

[ee] 


dial = 425 

busy = 425/300,0/300 

ring = 425/1000,0/4000 

congestion = 425/200,0/200 

callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600 

dialrecall = 425/650,0/25 

record = 1400/500,0/15000 

info = 950/650,0/325,950/325,0/30,1400/1300,0/2600 

stutter = 

!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/10 

0,!0/100,425 

[es] 


dial = 425 

busy = 425/200,0/200 

ring = 425/1500,0/3000 

congestion = 425/200,0/200,425/200,0/200,425/200,0/600 

callwaiting = 425/175,0/175,425/175,0/3500 

dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 

record = 1400/500,0/15000 

info = 950/330,0/1000 

dialout = 500 

[fi] 


dial = 425 

busy = 425/300,0/300 

ring = 425/1000,0/4000 

congestion = 425/200,0/200 

callwaiting = 425/150,0/150,425/150,0/8000 


dialrecall = 425/650,0/25 

record = 1400/500,0/15000 

info = 950/650,0/325,950/325,0/30,1400/1300,0/2600 

stutter = 425/650,0/25 

[fr] 


dial = 440 

busy = 440/500,0/500 

ring = 440/1500,0/3500 

congestion = 440/250,0/250 

callwait = 440/300,0/10000 

dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330 

stutter = 

!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/10 

0,!0/100,440 

[gr] 


dial = 425/200,0/300,425/700,0/800 

busy = 425/300,0/300 

ring = 425/1000,0/4000 

congestion = 425/200,0/200 

callwaiting = 425/150,0/150,425/150,0/8000 

dialrecall = 425/650,0/25 

record = 1400/400,0/15000 

info = 

!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1 

400/330,!1800/330,!0/1000,0 

stutter = 425/650,0/25 

[hk] 

ringcadence = 400,200,400,2000 

dial = 350+440 

busy = 480+620/500,0/500 

ring = 440+480/400,0/200,440+480/400,0/2000 

congestion = 480+620/250,0/250 

callwaiting = 440/100,0/4000 

dialrecall = 300/1500,0/500 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330 

[hu] 


dial = 425 

busy = 425/300,0/300 

ring = 425/1250,0/3750 

congestion = 425/300,0/300 

callwaiting = 425/40,0/1960 


record = 1400/400,0/15000 

info = 

!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1 

400/330,!1800/330,!0/1000,0 


stutter = 350+375+400 

[il] 

; Source: libtonezone. 

description = Israel 


dial = 414 

busy = 414/500,0/500 

ring = 414/1000,0/3000 

congestion = 414/250,0/250 

callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000 

dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414 

record = 1400/500,0/15000 

info = 1000/330,1400/330,1800/330,0/1000 

stutter = 

!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/16 

0,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414 

[in] 


description = India 

ringcadence = 400,200,400,2000 

dial = 400*25 

busy = 400/750,0/750 

ring = 400*25/400,0/200,400*25/400,0/2000 

congestion = 400/250,0/250 

callwaiting = 400/200,0/100,400/200,0/7500 

dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330,0/1000 

stutter = 

!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!35 

0+440/100,!0/100,!350+440/100,!0/100,350+440 

[it] 


dial = 425/200,0/200,425/600,0/1000 

busy = 425/500,0/500 

ring = 425/1000,0/4000 

congestion = 425/200,0/200 

callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000 

dialrecall = 470/400,425/400 

record = 1400/400,0/15000 

info = 

!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1 

400/330,!1800/330,!0/1000,0 

stutter = 470/400,425/400 

[jp] 


description = Japan 


dial = 400 

busy = 400/500,0/500 

ring = 400+15/1000,0/2000 


congestion = 400/500,0/500 

callwaiting = 400+16/500,0/8000 

dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330,0 

stutter = 

!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/10 

0,!0/100,400 

[lt] 


dial = 425 

busy = 425/350,0/350 

ring = 425/1000,0/4000 

congestion = 425/200,0/200 

callwaiting = 425/150,0/150,425/150,0/4000 

dialrecall = 425/500,0/50 

record = 1400/500,0/15000 

info = 

!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1 

400/330,!1800/330,!0/1000,0 

stutter = 

!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/10 

0,!0/100,425 

[mx] 


dial = 425 

busy = 425/250,0/250 

ring = 425/1000,0/4000 

congestion = 425/250,0/250 

callwaiting = 425/200,0/600,425/200,0/10000 

dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 

record = 1400/500,0/15000 

info = 950/330,0/30,1400/330,0/30,1800/330,0/1000 

stutter = 

!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!35 

0+440/100,!0/100,!350+440/100,!0/100,350+440 

[nl] 


dial = 425 

busy = 425/500,0/500 

ring = 425/1000,0/4000 

congestion = 425/250,0/250 

callwaiting = 425/500,0/9500 

dialrecall = 425/500,0/50 

record = 1400/500,0/15000 

info = 950/330,1400/330,1800/330,0/1000 

stutter = 425/500,0/50 

[no] 


dial = 425 

busy = 425/500,0/500 

ring = 425/1000,0/4000 


congestion = 425/200,0/200 

callwaiting = 425/200,0/600,425/200,0/10000 

dialrecall = 470/400,425/400 

record = 1400/400,0/15000 

info = 

!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1 

400/330,!1800/330,!0/1000,0 

stutter = 470/400,425/400 

[nz] 

ringcadence = 400,200,400,2000 

dial = 400 

busy = 400/250,0/250 

ring = 400+450/400,0/200,400+450/400,0/2000 

congestion = 400/375,0/375 

callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200 

dialrecall = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,400 

record = 1400/425,0/15000 

info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400 

stutter = 

!400/100!0/100,!400/100,!0/100,!400/100,!0/100,!400/100!0/100,!400/100,!0/100,!400/100, 

!0/100,400 

[pl] 


dial = 425 

busy = 425/500,0/500 

ring = 425/1000,0/4000 

congestion = 425/500,0/500 

callwaiting = 425/150,0/150,425/150,0/4000 

dialrecall = 425/500,0/50 

record = 1400/500,0/15000 

info = 

!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1 

400/330,!1800/330,!0/1000 

stutter = 

!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/10 

0,!0/100,425 

[pt] 


dial = 425 

busy = 425/500,0/500 

ring = 425/1000,0/5000 

congestion = 425/200,0/200 

callwaiting = 440/300,0/10000 

dialrecall = 425/1000,0/200 

record = 1400/500,0/15000 

info = 950/330,1400/330,1800/330,0/1000 

stutter = 

!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/10 

0,!0/100,425 

[ru] 


dial = 425 


usy = 425/350,0/350 

ring = 425/800,0/3200 

congestion = 425/350,0/350 

callwaiting = 425/200,0/5000 

dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330,0 

[se] 


dial = 425 

busy = 425/250,0/250 

ring = 425/1000,0/5000 

congestion = 425/250,0/750 

callwaiting = 425/200,0/500,425/200,0/9100 

dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 

record = 1400/500,0/15000 

info = 

!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/3 

32,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/ 

24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0 

stutter = 

!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/10 

0,!0/100,425 

[sg] 

ringcadence = 400,200,400,2000 

dial = 425 

ring = 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90% 

busy = 425/750,0/750 

congestion = 425/250,0/250 

callwaiting = 425*24/300,0/200,425*24/300,0/3200 

stutter = 

!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/60 

0,!0/200,!425/200,!0/200,!425/600,!0/200,425 

info = 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference 

dialrecall = 425*24/500,0/500,425/500,0/2500 ; unspecified in IDA reference, use repeating 

Holding Tone A,B 

record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s 

nutone = 425/2500,0/500 

intrusion = 425/250,0/2000 

warning = 425/624,0/4376 ; end of period tone, warning 

acceptance = 425/125,0/125 

holdinga = !425*24/500,!0/500 ; followed by holdingb 

holdingb = !425/500,!0/2500 

[uk] 

ringcadence = 400,200,400,2000 

dial = 350+440 

specialdial = 350+440/750,440/750 

busy = 400/375,0/375 

congestion = 400/400,0/350,400/225,0/525 

specialcongestion = 400/200,1004/300 

unobtainable = 400 

ring = 400+450/400,0/200,400+450/400,0/2000 

callwaiting = 400/100,0/4000 

specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000 

creditexpired = 400/125,0/125 

confirm = 1400 

switching = 400/200,0/400,400/2000,0/400 


info = 950/330,0/15,1400/330,0/15,1800/330,0/1000 

record = 1400/500,0/60000 

stutter = 350+440/750,440/750 

[us] 


dial = 350+440 

busy = 480+620/500,0/500 

ring = 440+480/2000,0/4000 

congestion = 480+620/250,0/250 

callwaiting = 440/300,0/10000 

dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330,0 

stutter = 

!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!35 

0+440/100,!0/100,!350+440/100,!0/100,350+440 

[us-old] 


dial = 600*120 

busy = 500*100/500,0/500 

ring = 420*40/2000,0/4000 

congestion = 500*100/250,0/250 

callwaiting = 440/300,0/10000 

dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330,0 

stutter = 

!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600* 

120/100,!0/100,!600*120/100,!0/100,600*120 

[tw] 


dial = 350+440 

busy = 480+620/500,0/500 

ring = 440+480/1000,0/2000 

congestion = 480+620/250,0/250 

callwaiting = 350+440/250,0/250,350+440/250,0/3250 

dialrecall = 300/1500,0/500 

record = 1400/500,0/15000 

info = !950/330,!1400/330,!1800/330,0 

stutter = 

!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!35 

0+440/100,!0/100,!350+440/100,!0/100,350+440 


APPENDIX B 

B.1 – AUSTRALIAN VSPS (TRUNKS) 

The following is a collection of TRUNK settings recommended for other VSPs that are not 

covered above. Details of trunks configuration are correct at the time they were compiled, 

however changes may have been made. Users are advised to consult their respective 

VSPs for confirmation. 

Note: In Peer Details, insert qualify=yes if you want fail-over to PSTN should you loose 

your internet connection. 

B.1.1 BBPglobal 

Peer Details 

Trunk Name: bbpglobal 

allow=g723&gsm 

authuser= 

disallow=all 

fromdomain=sip2.bbpglobal.com 

fromuser= 

host= sip2.bbpglobal.com 

insecure=very 

secret= 

type=peer 

username= 

User Details 

User Context: 


fromuser= 

insecure=very 

secret= 

type=user 

username= 

Note:You may need to include the 

following in your sip.conf 

Maxexpirey=3600 

Qualify=yes 

Register String: 

:sip2.bbpglobal.com/ 

B.1.2 Engin BYO 

Peer Details 

Trunk Name: engin 


auth=md5 

canreinvite=yes 

disallow=all 


fromdomain=voice.mibroadband.com.au 

fromuser=02321XXXX 

host=byo.engin.com.au 

insecure=very 

musiconhold=framed 

nat=yes 

port=5060 

qualify=no 

realm=mobileinnovations.com.au 

reinvite=yes 

secret= 

type=friend 


User Details 

User Context: 02321XXXX 


fromdomain=voice.mibroadband.com.au 

host=byo.engin.com.au 

secret= 

type=user 


Note: 

You may need to include the following in 

your sip.conf 

Defaultexpirey=600 

Maxexpirey=3600 


02321XXXX:@byo.engin.com.au/02321XXXX 


B.1.3 Faktortel (IAX) 

Peer Details 

Trunk Name: faktortel 

allow=g729&gsm&ulaw&alaw&ilbc 

disallow=all 

host=iax.faktortel.com.au 

qualify=3000 

secret= 

type=friend 

username=xxxxxx 

User Details 

User Context: xxxxxx 

auth=md5 


host=iax.faktortel.com.au 

qualify=3000 

secret= 

type=friend 

username=xxxxxx 


xxxxxx:@iax.faktortel.com.au 

Faktortel supports the following codecs: ulaw, alaw, ilbc, gsm, g729. The following is 

the link on the discussions for setting up Faktortel on TRIXBOX. 

http://www.faktortel.com.au/support-Trixbox-settings.shtml 

B.1.4 iPrimus 

Primus is a strange animal. To get it to work with Trixbox requires me to do a bit of 

detective work. With my partner in crime, Colin, we set out to play detective because 

we need to obtain the correct credential to get Trixbox to register and to be able to 

make and receive calls through Trixbox. Primus Telecom has refused to assist. 

To get the necessary credentials, we cheated a little as the method was published in 

the whirlpool forum http://forums.whirlpool.net.au/forum-replies.cfmt=604015 

First of all, using a browser, we need to enter the following address: 

https://spconfig.voip.iprimus.net.au/SoftphoneConfigurationServer/Default.aspxAction=LOG 

IN&username=username&password=password&computerid=db7b351e57af&MAC=your 

_NIC_MAC_address 

Replace the bold type with your Talk Broadband username and password given to 

you by iPrimus. You can get your NIC MAC address by running ipconfig /all. 

When you submit that address, the following data will be displayed in your browser. 

[Data] Success=1 [Settings] proxies:proxy0:display_name="02********" 

proxies:proxy0:username="AU_********" proxies:proxy0:password="********" 

system:auto_update:config_server_url="https://spconfig.voip.iprimus.net.au/Soft 

phoneConfigurationServer/Default.aspxAction=CONFIG&Username=username&Version=1 

&MAC=$MAC$&computerid=$computerid$" audio:headset:audio_in_agc_enabled="0" 

audio:speaker_phone:audio_in_agc_enabled="0" codecs:speex_fec:enabled="0" 

codecs:speex_wb:other_bandwidths="24000,26000,28800,32800,40000,44000" 

codecs:speex_wb:starting_bandwidth="44000" 

codecs:speex_wb:usable_bandwidths="20800,24800,28800,36800,44000" 

codecs:speex_wb_fec:enabled="0" proxies:proxy0:contacts_etag="" 

proxies:proxy0:contacts_last_upload_failed="0" proxies:proxy0:contacts_mod="0" 

proxies:proxy0:default_session_refresh_interval_in_seconds="60" 

proxies:proxy0:digit_map="#1\a\a.T;match=1;prestrip=2;" 

proxies:proxy0:domain="asnjla2.bw.iprimus.net" 

proxies:proxy0:enable_session_timer="0" proxies:proxy0:enabled="1" 

proxies:proxy0:ice_enabled="1" 

proxies:proxy0:maximum_time_before_reregistration_in_seconds="1800" 

proxies:proxy0:minimum_time_before_reregistration_in_seconds="20" 

proxies:proxy0:override_outbound_proxy="1" 

proxies:proxy0:override_sip_listen_port="0" 

proxies:proxy0:override_stun_server="1" proxies:proxy0:port_range_enable="0" 

proxies:proxy0:proxy="sbc01a.syd.voip.iprimus.net.au:15060" 

proxies:proxy0:register="1" proxies:proxy0:reregister_in_seconds="3600" 


Note the bold highlights. These are the data that you need for your trunk setting. 

Armed with the data above, you may now set your trunk setting. 

Peer Details 

Trunk Name: iPrimus 

allow=alaw&ulaw 

disallow=all 

fromdomain=asnjla2.bw.iprimus.net 

fromuser=AU_XXXXXXXX 

host=asnjla2.bw.iprimus.net 

insecure=very 

outboundproxy=sbc01a.syd.voip.iprimus.net.au 

port=15060 

secret=YYYYYYYY 

type=peer 

username=AU_XXXXXXXX 

User Details 

User Context: 02******** 


type=user 


AU_XXXXXXXX@asnjla2.bw.iprimus.net:YYYYYYYY:AU_XXXXXXXX@iprimus/02******** 

B.1.5 Koala 

Peer Details 

Trunk Name: KoalaSip 

allow=g729&gsm&alaw&ulaw 

disallow=all 

fromuser= 

host=203.122.248.173 

nat=yes 

port=5060 

qualify=no 

secret= 

type=friend 

User Details 



fromdomain=203.122.248.173 

host=203.122.248.173 

secret= 

type=user 

username= 


:@koalavoip.com.au 

(Source Ref: Curtis of Koala) also refer here https://www.koalavoip.com.au/billing/node/75 

B.1.6 MyNetFone 

Peer Details 

Trunk Name: MyFonesip 

allow=g729&alaw&ulaw 

authname=091xxxxx 


disallow=all 



host=sip01.mynetfone.com.au 

insecure=very 

nat=yes 

pedantic=no 

qualify=yes 

secret= 

type=friend 

username=091xxxxx 

User Details 

User Context: 091XXXXX 




insecure=very 

qualify=no 

secret= 

type=friend 




091xxxxx@sip01.mynetfone.com.au::091xxxxx@sip01.mynetfone.com.au/091xxxxx 

For further information on MyNetFone, you may refer to the following link: 

http://www.myfone.com.au/downloads/sip.conf 

http://www.myfone.com.au/downloads/extensions.conf 

B.1.7 Nehos (IAX) 

Peer Details 

Trunk Name: Nehos 

allow=g729 

disallow=all 

host=iax.ifone.com.au 

qualify=yes 

secret= 

type=peer 

username=661xxxx 

User Details 

User Context: 661XXXX 


host=iax.ifone.com.au 

secret= 

type=user 



661xxxx:@iax.ifone.com.au 

B.1.8 Nodephone 

Peer Details 

Trunk Name: Nodephone 

allow=g729 


disallow=all 


fromdomain=sip.internode.on.net 

fromuser= 

username= 

host=sip.internode.on.net 

insecure=very 

secret= 

type=peer 

User Details 



host= sip.internode.on.net 

secret= 

type=user 

username= 


:@sip.internode.on.net/ 

B.1.9 SipMe 

Peer Details 

Trunk Name: sipme 

authname=1777xxxxxx 


fromuser=1777xxxxxx 

host=sip.sipme.com.au 

insecure=very 

secret= 

type=peer 

username=1777xxxxxx 

User Details 

User Context: 1777xxxxxx 


fromuser=1777xxxxxx 

secret= 

type=user 


1777xxxxxx:@sip.sipme.com.au/1777xxxxxx 


B.2 – NEW ZEALAND VSPS (TRUNKS) 

B.2.1 iTalk 

Peer Details 

Trunk Name: italk 

allow=ulaw&g729 



disallow=all 



host=akl.italk.co.nz 

insecure=very 

secret= 

type=friend 


User Details 

User Context: 

You may prefix outgoing 

calls with 0197 to disable 

outgoing CID. 

Inbound route 64997xxxxx 

required to be forwarded 

to a destination ext/menu 


64997xxxxx:@akl.italk.co.nz/64997xxxxx 

(Source Ref: Steve Biddle) 

B.2.2 Double Zero 64 (IAX) 

Peer Details 

Trunk Name: 0064 

allow=ulaw&g729 

disallow=all 


host=voip.0064.co.nz 

insecure=very 

secret= 

type=peer 

username=your_Customer_ID 

User Details 

User Context:; Your_Customer_ID 


host=voip.0064.co.nz 

insecure=very 

secret=PASSWORD 

type=user 

username=CUSTOMER_ID 


Your_Customer_ID: @voip.0064.co.nz 


B.3 – OTHER VSPS (TRUNKS) 

B.3.1 Broadvoice 

Peer Details 

Trunk Name: Broadvoice 

allow=ulaw 


disallow=all 


fromdomain=sip.broadvoice.com 

fromuser= 

host=sip.broadvoice.com 

insecure=very 

nat=yes 

secret= 

type=peer 

username= 

User Details 



type=user 


@sip.broadvoice.com::@sip.broadvoice.com/ 

B.3.2 FWD aka Pulver(IAX) 

Peer Details 

Trunk Name: fwd 


auth=rsa 


disallow=all 

host=iax2.fwdnet.net 

inkeys=freeworlddialup 

insecure=very 

secret= 

type=peer 


User Details 

User Context: 65xxxx 

auth=rsa 


inkeys=freeworlddialup 

type=user 


65xxxx:@iax2.fwdnet.net 

B.3.3 SipBroker 

Peer Details 

Trunk Name: sipbroker-out 

allow=g729&ulaw&alaw 


disallow=all 


fromdomain= 

fromuser= 

host=sipbroker.com 

insecure=very 

nat=yes 

port=5060 

secret= 

type=peer 

You will not require 

user details since you 

will not be receiving 

incoming calls from this 

trunk. 

You do not need to 

register either. 


If you are using SipBroker, you must ensure that your sip_nat.conf is modified to the 

following; 

nat=yes 

externip= ; if you have fix IP or 

externhost= ; if you are using Dynamic IP 

localnet=192.168.1.0/255.255.255.0 

B.3.4 SipPhone 

Peer Details 

Trunk Name: sipphone 

fromdomain=proxy01.sipphone.com 

host=proxy01.sipphone.com 

insecure=very 

secret= 

type=peer 

username=1747xxxxxxx 

User Details 

User Context: 1747xxxxxxx 



fromuser=1747xxxxxxx 

insecure=very 

qualify=no 

secret= 

type=user 

username=1747xxxxxxx 


1747xxxxxxx:@proxy01.sipphone.com/1747xxxxxxx 

B.3.5 VOIP Buster 

Peer Details 

Trunk Name: VoipBuster 

allow=alaw&ulaw 


disallow=all 

dtmfmode=inband 

fromdomain=sip1.voipbuster.com 

fromuser= 

host=sip1.voipbuster.com 

insecure=very 

nat=yes (necessary in some 

cases) 

qualify=yes 

secret= 

srvlookup=yes 

type=friend 

username= 

User Details 

User Conext: 

If you have an inbound number, an 

Inbound route is required to forwarded 

your incoming call to a destination 

extension or IVR menu: 

Inbound Route: 

DID: 

No Fax 


:@sip1.voipbuster.com/ 

(Source Ref: Florent Chandelier) 


APPENDIX C 

Details for Asterisk Sound Files can be obtained from the following links: 

http://www.voip-info.org/tiki-index.phppage=Asterisk+sound+files 

http://www.voip-info.org/wiki/view/Asterisk+sound+files+additional 

http://www.nathanpralle.com/software/ast_masterlist.html 

C.1 VOICE FOR ASTERISK MESSAGES 

Brian, a Whirlpool Enthusiast, posted the following tip in the Whirlpool forum. Brian has 

asked me to include his discovery in this guide. 

The following is an extract of Brian’s exploits – verbatim ☺ 

 

I have just been setting up the Digital Receptionist and I wanted to use a professional 

voice for messages. I thought I might detail my exploits here for those that might be 

considering the same thing. 

First you need the original recording either from a real professional or a "text to voice" 

utility. I went with the Realspeak demo 

http://www.nuance.com/realspeak/demo/default.asp 

The "British Daniel" voice was deep and professional. Check this out 

http://www.cbservers.com.au/whirlpool.wav 

Now the demo has a 100-character limit so you may need to do 2 or 3 files and join them 

together with a program like Accoustica MP3 mixer. 

http://www.acoustica.com/mp3%2Daudio%2Dmixer/ 

Save it as a wav file. 

It was after this part that I had a lot of problems. Asterisk is very funny about sound files. I 

tried using all sorts of programs to get the file to the right specs but it failed to work every 

time. I eventually used "sox" an open source command line editing tool and it worked first 

go (using advice from VoIP-info & digium forums) http://sox.sourceforge.net/ then at the 

command line prompt use this syntax: 

sox winwave.wav -r 8000 -c 1 -s -w linwave.wav 

you may then upload your message to asterisk. 

Some further reading; 

http://lists.digium.com/pipermail/asterisk-users/2002-December/006635.html 

http://www.voip-info.org/tiki-index.phppage=Convert+WAV+audio+files+for+use+in+Asterisk 

http://users.pandora.be/Asterisk-PBX/IVR.htm 

http://www.voip-info.org/tiki-index.phppage=Asterisk+sound+files 

 


C.2 AUSTRALIAN VOICE AND LANGUAGE SET FOR ASTERISK 

VOIPSHOP has made available the files free to non-commercial users. Log on to 

http://www.voipshop.com.au/product_info.phpproducts_id=72 to download 

The following is extract of the instructions by VOIPSHOP on their web site. Two files are 

available; one is as a replacement for the current asterisk files. The other is as an 

additional Asterisk language (if you are going to use the SetLanguage option). 

Replace Current Asterisk Files: ausounds.tar.gz 

Quick Steps for TRIXBOX. 

Log into your TRIXBOX server and at the command prompt, do the following: 

wget http://files.voipshop.com.au/ausounds.tar.gz 

tar cvfz backupsounds.tar.gz /var/lib/asterisk/sounds/ (for backing 

up your existing files) 

tar -zxvf ausounds.tar.gz -C /var/lib/asterisk/sounds/ 

Done. Your asterisk sound files are replaced. Make sure you back up, just in case. 

Additional Asterisk Language Set: ausoundset.tar.gz 

At the command prompt execute the following commands: 

wget http://files.voipshop.com.au/ausoundset.tar.gz 

tar -zxvf ausoundset.tar.gz -C /var/lib/asterisk/sounds/ 

Done. Your asterisk sound files now contain an au soundset. 

Note : You can only download these files from voipshop, 

Please do not re-distribute. These are free for non-commercial use only. Contact 

VOIPSHOP mailto:support@voipshop.com.au for commercial license details. 

C.3 MORE AUSTRALIAN VOICE FOR ASTERISK 

As part of OpenVoice contribution to the Asterisk community, they are offering a free set 

of Australian prompts for the Asterisk Open Source PBX. 

The files (male voice) can be downloaded here: http://www.openvoice.com.au/free.php 

Of course, don’t forget to add the following line in the iax.conf and sip.conf configuration 

file under [general]. In zapata.conf configuration file the line should be added under 

[channels]. 

language=au ; (or fr, it or whichever language is installed) 


APPENDIX D (ATAS AND ROUTERS) 

D.1 ALCATEL SPEEDTOUCH 510/530 ADSL ROUTER 

If you are using Telstra Alcatel SpeedTouch 510/530 ADSL Router, you will need to 

unbind port 5060, otherwise you will not be able to use VOIP since Alcatel, in their 

wisdom for reasons best known to themselves, have decided that we are not suppose to 

use this port. So they blocked it in these types of routers. 

But, as we know, there are more than just one way to skin a cat ☺ thus, the solution is to 

telnet to the ADSL modem and remove this restriction. 

I am assuming that you are running a Windows PC to do this. 

Click on Start > Run then type: 

telnet 10.0.0.138 [or replace with the correct ip address] 

you will notice the following; 

Trying 10.0.0.138... 

Connected to speedtouch 

Escape character is '^]'. 

Username : root 

Password : 

---------- 

... 

... welcome message blah blah blah 

... 

You then should unbind the SIP protocol and reboot the modem thus: 

=>nat unbind application=SIP port=5060 

=>config save 

=>system reboot 

To check whether the SIP protocol is bound in your modem use this command to show 

the list of bindings (only showing SIP here): 

=>nat bindlist 

Application Proto Port 

SIP udp 5060 

If the line doesn't appear, your phone should work. 

Source Reference: 

http://www.sipgate.co.uk/faq/index.phpaktion=artikel&type=faq&rubrik=024&id=398&lang=de 

D.2 BILLION 7402 VL 

Some extra information that will get a Billion 7402 VL FXS to work with asterisk and have 

NAT port 5060 points to the Asterisk server. 

By default, if you enable the FXS ports on the Billion 7402 VL router to be sip enabled, 

the router automatically forwards the port (5060) to its own IP address and an error will 

occur if you try to forward that port to the Asterisk box. This is because it is intended for 

the FXS ports to connect directly to an external SIP provider. 


To overcome this issue and make the FXS ports connect to the Asterisk box, complete 

the following: 

From a windows command prompt… 

telnet 192.168.XXX.XXX 

(router I.P. address) 

voip set interfacename iplan (this changes it from the WAN to the LAN interface 

Next, power reset the router. 

Now you can add the NAT to point to the asterisk server and the FXS ports will connect to 

the Asterisk box. 

NOTE: - Also codec G729 had to be disabled on the FXS ports for calls to work. Not sure 

why. 

(Source Reference: Sean Mahon) 

D.3 LINKSYS PAP2 AND SIPURA SPA2000 

The most common Analog Telephone Adapters (ATA) amongst Australian VOIP users 

are the Linksys PAP2, Sipura 2000 and Sipura SPA3000 (the Sipura SPA3000 was 

covered in the chapter relating to Interfacing Asterisk to PSTN). The Linksys PAP2 and 

the Sipura SPA2000 are essentially identical units encased in different boxes and 

branded differently. The examples presented here apply to both the Linksys PAP2 and 

Sipura SPA2000. 

Note: 

The set-up described here is for use with TRIXBOX and not for connecting the unit 

directly to the VOIP service provider. 

I will describe the minimum changes that are required to get these ATAs to work in 

conjunction with Asterisk. 

As supplied, the unit comes with DHCP enabled. It is best to allocate a fix IP address to 

the unit and therefore the first thing to do is to disable the DHCP and allocate a fix IP 

address e.g. 192.168.1.50. You need to do the following: 

1. Connect the unit to your network via the RJ45 port behind the unit. 

2. Power up the unit 

3. Plug in a telephone in the RJ11 port 1 behind the unit (any one will do) 

4. Dial **** and listen to the instruction that you will hear from the IVR 

5. Dial 101 and when prompted enter 0 followed by # (this will disable DHCP) 

6. Dial 111 and when prompted enter 192*168*1*50 followed by # (assuming that 

you want to allocate 192.168.1.50 as the IP address of the ATA). 

7. Dial 120 and you will hear the current sub-net mask. If it is not your existing 

network subnet mask you may want to change it. If it is the same as your current 

network’s subnet mask, do don’t need to do anything else). If you need to 

change the sub-net mask, do the following: 

8. Dial 121 and at the prompt enter 255*255*255*0 (assuming that is your existing 

sub-net mask) 

Next, connect to the Unit using your browser by entering http://192.168.1.50 in your 

browser address bar and hit enter. 


2. Select Admin Login and then select (switch to advanced view) 

3. Select the System Tab 

a. Under Internet Connection Type, change the Gateway address to the 

address of your router. 

4. Select Regional Tab 

a. Under Vertical Service Activation Codes, delete all the fields. 

5. Select Line 1 Tab 

a. Under SIP Setting, change the SIP Port to 5060 

b. Under Proxy and Registration, change the Proxy to reflect the address 

of your Asterisk box. 

c. Under Subscriber Information, change the Display Name to whatever 

you like, change the User ID to the Asterisk Extension number that you 

have allocated for this line e.g. 201. Change the Password to the 

password you have allocated to the asterisk extension. Change Use 

Auth ID to Yes. Change the Auth ID to the extension number you have 

allocated e.g. 201, the same as the User ID. 

d. Under Dial Plan, change the existing dial plan to (**|*x.|x.|**x.) Change 

Enable IP Dialling to yes. 

6. Select Line 2 Tab 

a. Under SIP Setting, change the SIP Port to 5061 

b. Under Proxy and Registration, change the Proxy to reflect the address 

of your Asterisk box. 

c. Under Subscriber Information, change the Display Name to whatever 

you like, change the User ID to the Asterisk Extension number that you 

have allocated for this line e.g. 202. Change the Password to the 

password you have allocated to the asterisk extension. Change Use 

Auth ID to Yes. Change the Auth ID to the extension number you have 

allocated e.g. 202, the same as the User ID. 

d. Under Dial Plan, change the existing dial plan to (**|*x.|x.|**x.) Change 

Enable IP Dialling to yes. 

7. Set this setting only if you are using STUN Server 

a. Select SIP Tab 

b. Handle VIA received: no 

c. Handle VIA rport: no 

d. Insert VIA received: no 

e. Insert VIA rport: no 

f. Substitute VIA Addr: yes 

g. Send Resp To Src Port: yes 

h. STUN Enable: yes 

i. STUN Test Enable: no 

j. STUN Server: stun.xten.com (I use stun.xten.com, you may use 

whichever stun server you like) 

k. NAT Keep Alive: 15 

These are the minimum changes you need to do to get the ATA working in conjunction 

with Asterisk. There are other changes that you can do depending on your own 

requirement. There are various discussions on this topic in the Whirlpool, Voxilla and 

other forums that you can refer to. 


APPENDIX E (USERS’ CONTRIBUTIONS) 

E.1 SUGGESTIONS FOR DIAL PLAN AND CONFIGURATION 

Place Holder for Suggestions on Dial Plan and Configurations 


E.2 MULTIPLE TDM400P INSTALLATION 

Neilmc, a participant I the Whirlpool Forum provided the following feedback on the use of 

multiple TDM400P Digium cards. 

This is being discussed in this thread: 

http://forums.whirlpool.net.au/forum-replies.cfmt=483550&p=1 

 

There are plenty of forum posts (on WP and elsewhere) + various how-to and guides that 

say don't use more than one if you want a reliable system with no audio problems. 

I'm sure that on the old Digium site they had a caution not to use more than one TDM or 

TE card (That's changed, they now brag about 1 box running 5 quad span TE cards). 

This machine is now working in a busy environment (medical centre) nicely, sometimes 

under full Zap call load. It has 5 x FXO & 5 x FXS on 3 cards. CPU is under 10% pretty 

much all the time. 

I asked them to be very fussy about audio quality and let me know if they have any echo, 

pops, clicks, distortion etc. 

So far everyone is saying that it's perfectly fine, not a bad call yet. 

There is good reason behind the old advice not to do it though. Lots of people have had 

grief. 

Make sure you don't grab any old motherboard that is lying about (especially if it isn't PCI 

2.2 compliant) Even though minimum system specs aren't high don't go for a bottom of 

the range el-crapo brand motherboard. There is a list of some incompatible motherboards 

on the Digium site, but no doubt there will be others that have problems. 

Choose a board that has plenty of PCI slots (5 or so). Make sure that it has plenty of 

control over IRQ in CMOS. Eg the ASUS P5P-800 I used could use APIC to assign IRQ 

or you could manually assign an IRQ to a particular PCI slot. 

Be very sure that each Digium card is not sharing an IRQ with anything else. 

Disable any onboard devices you don't need inc. serial, parallel & USB ports. 

It may not be a problem, but to minimise chances of problems disable Hyper threading if it 

is supported on your CPU. Use 32 bit OS rather than 64 bit. 

Use plenty of RAM. You don't want a PABX to be thrashing about with a swap file. I used 

1GB of decent quality RAM which is definitely far more than the system needs. (It's using 

about 230MB at the moment), but 1GB of RAM doesn't really cost much any more. 

Check that your hard drives are running in DMA mode (or use SCSI drives). Sometimes 

they default to PIO, which might cause problems if there is a sudden burst of disk activity. 

Sometimes APIC can cause you grief. You might be able to tweak your kernel, but you 

might get what you need by turning it off. 

Have a read through 

http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html 

If you have all of the driver stuff loading properly and still have audio problems. 

Use recent zaptel drivers (unless you become aware of an issue with them). 


• Make sure that the driver loads with the module option opermode=AUSTRALIA. 

• Run fxotune -i 4 (usually from /usr/src/zaptel) It will create a file /etc/fxotune.conf 

with settings for your cards based on tests with your lines (takes about 5 minutes 

per FXO port). 

• You will have to stop Asterisk before running fxotune. 

• Have /usr/src/zaptel/fxotune -s run at startup to have the card set. Apparently the 

values in the conf file are often zero if you have the modules in the correct 

opermode. ( have one module at the office where the first value is 10 rather than 

zero. I'll try fxotune again and see if it changes at all). 

I just put /usr/src/zaptel/fxotune -s at the bottom of /etc/rc.d/rc.local in A@H 2.6 and it 

loaded okay. You need it to run after the drivers load but before asterisk starts. 

• Check /var/log/messages after boot up. Each FXO port should have loaded with 

AUSTRALIA mode. If it is FCC mode then the card is set for North American 

lines. 

• Each FXS port should have an entry for boosting ringer. 

If fxotune is loading settings okay you should see a line saying something like after all of 

the other init stuff. 

kernel: -- Setting echo registers: 

kernel: -- Set echo registers successfully 

If you have 4 FXO ports, you should see this 4 times. 

 


E.3 ELIMINATING ECHO PROBLEMS IN SPA-3000 

Tony, another participant in the Whirlpool Forum, provided the following recommendation. 

Echo in the SPA-3000 is a common problem. In reality, most of the time the SPA-3000 

isn't causing the echo, it's just making it more noticeable. This is due to the fact that the 

SPA-3000 passes calls from the PSTN to the LINE1 by converting it to VoIP internally 

and then backs to analogue. This process does not produce any echo, however is can 

add about 30ms of latency to the call. This added latency can make echo that was 

previously unnoticed suddenly become annoying. A pure digital system has no echo (the 

TX and RX path are 100% separated). It's the interaction of the Digital and Analogue that 

cause problems. 

This may help getting rid of that pesky echo on your Sipura SPA-3000 PSTN line: 

1. Make sure you are running the latest firmware (3.1.7) and you have everything back to 

factory defaults or at least undo all the previous tweaking. 

2. Switch *off* all echo can in all your devices. There are 6 places in the sipura to switch 

off echo can. 

• PSTN Line -> "Echo Canc Enable", 

• "Echo Canc Adaptive Enable", 

• "Echo Supp Enable" and 

• Line 1 -> "Echo Canc Enable", 

• "Echo Canc Adaptive Enable", 

• "Echo Supp Enable". 

The idea is that we want to hear how bad the echo is with different configs. 

3. Unplug everything from your phone line except the SPA-3000. This includes all the 

extension cables even with nothing connected to them. These can cause impedance 

problems that lead to echo. 

4. Set the Impedance on your lines. 

• PSTN -> "Port Impedance" = 220+820||120nF as a starting point. 

• Regional -> FXS Port Impedance = "220+820||115nF" as a starting point. 

5. In the PSTN tab set – 

• "Tip/Ring Voltage Adjust: = 3.1V" and 

• "Operational Loop Current Min = 16mA". 

Doesn't seem to affect echo, but I believe that these are the correct numbers for 

Australia. 

6. Turn down the jitter buffers! 

• "PSTN -> Network Jitter Level: = low", 

• "PSTN -> Jitter Buffer Adjustment: = disable". 

This reduces the delay across your SPA-3000. 

• "LINE1 -> Network Jitter Level: = low", 

• "LINE1 -> Jitter Buffer Adjustment: = up and down". 

If you are using a poor quality VoIP service as well as the PSTN then you could change 

the 

• "LINE1 -> Network Jitter Level: = medium". 

7. Set the preferred codec for the PSTN to be g711a and lock it in. 

• "PSTN -> Preferred Codec = g711a", 

• "PSTN -> Use Pref Codec Only = yes". 


Obviously adjust this if you’re accessing your PSTN line via VoIP from a remote network. 

Set the LINE1 to allow g711a as well as whatever else your prefer. 

• "LINE1 -> Use Pref Codec Only = no". 

The g711a is fast to encode and decode. Using this codec again reduces your latency 

and may make the echo less obvious or easier to catch with the echo canceller. 

8. Power cycle the SPA-3000 (pull the power plug). Believe it or not, this sometimes fixes 

the problem. Especially after you have changed the physical phone wiring. 

9. Make some test calls. The telco test number 1800801920 is a good one to start with. It 

has a recorded voice telling you your local phone number. While it's talking, talk back and 

work out how much echo you are getting. Talk loud, talk soft. 

10. Look at what you have got. If you can hear an echo then the problem could be that 

your probably sending to much power down the line. This is probably reflecting back 

somewhere as an echo. If you’re close to the exchange and have good wires then this is 

probably the case. You need to crank back on the power. Go to PSTN -> "SPA To PSTN 

Gain" and turn it down. Be aware that at some point if you turn it down to much, the SPA 

sorts does a double negative and turns it way up. I believe the range of this variable is 

about -127 -> 127 (from my testing). Turn it down, down, down, down until the person can 

still hear you but reduced echo. 

Note: if you enable "Echo Supp Enable" then you will negate these parameters. It seems 

that the Sipura echo suppression is actually just an automatic gain control. It's really 

annoying - leave it off. 

11. Make a test call to someone with a known good phone out via the SPA's PSTN line or 

get someone to call in to the PSTN line. Best if it’s just a boring old Telstra phone hard 

wired to a socket on the wall. Don't call a mobile! 

If the remote party is hearing echo, it could be that your phone is so loud that it's feeding 

back into the microphone. Turn down the PSTN To SPA Gain until you can comfortable 

hear the person, no more. If the remote user can still hear echo, try using a different 

phone plugged into the SPA. Go for the basics first, a cruddy old Telstra phone is what I 

use for testing. If this solves the problem you may have a bad phone or an impedance 

miss match between your phone and the SPA. 

• Try changing the Regional -> FXS Port Impedance to "600". 

• Try changing the FXO port impedance to "600" or "global". 

If this doesn't help, change it back. The impedance will only affect what the *remote* party 

hears, it won't help echo you are hearing. 

12. After you have the echo down to a reasonable level, go back into the "PSTN" tab and 

switch on the "Echo Can Enable = yes". Check to see if the echo has improved. If the 

echo is tolerable at this level, leave the adaptive echo canceller off. You should have the 

echo level down to a level that can be stomped on by the echo canceller. If you are using 

a sip device to talk through your PSTN line, you should probably do all the echo 

cancellation at that device and leave it switched off in the SPA. 

The adaptive echo canceller is a lot more aggressive but also can cancel out some of the 

incoming conversation. In particular if you’re calling in a loud environment then the voice 

going down the line from your end can trick the echo canceller to start canning some of 

the real conversation. It makes the incoming party sound a bit scratchy. Leave it off 

unless you really need it. 

The "Echo Supp Enable" switches on automatic gain controls. This means the Sipura will 

be constantly turning up and down the volume of the call for you and the remote party to 


try and keep the sound levels at a "good" level. Sipura's definition of good may not suit 

your situation. The constant variation of the volume annoys me so I recommend leaving it 

"off". 

13. Test call. Talk loud, talk soft. Is your conversation clear at both ends Should be a lot 

better. If echo re-appears, at a later date, remember to try power cycling the SPA before 

you tweak with anything. It may also be at the other end of the call. 

In the end it basically boils down to this – 

If you are hearing the echo, the real problem is at the *other* end of the connection. In a 

normal phone conversation, the latency is so low that you don't notice it. Your brain 

automatically tunes out to its own voice when you’re talking (as long as the delay between 

talking and hearing isn't to long). When you insert the SPA, the delay is increased and 

sometimes the echo can become noticeable. 

The only way you can counter this is to reduce the latency (jitter buffer) and/or reduce the 

gains so the echo volume is reduced below the level that it is noticed. At this point, the 

SPA-3000 echo canceller should be able to kill off the rest. 

E.4 IMMEDIATE ANSWER OF PSTN CALLS 

By default, calls will be answered after about 3 rings thus allowing time for Asterisk to 

detect the CID. Some users may want the PSTN calls to be answered by Asterisk 

immediately especially if there is a common phone connected in parallel and to avoid 

someone answering the call using the parallel phone. 

To do this you need to change one setting in the zapata.conf file. By default the setting is 

set to: 

Immediate = no 

To force Asterisk to answer immediately, change the line to read: 

Immediate = yes 

If you are not using Fax, it is also a good idea to disable fax extension in your General 

setting. 

This may or may not work satisfactorily as Asterisk needs a couple of rings to detect CID. 

E.5 REBUILD ZAPTEL DRIVER (USERS’ SUGGESTIONS) 

E.5.1 As experienced by marner – a Whirlpool Forum participant. 

On his Dell box, marner had to do the following to rebuild the driver and for it to work. 

Log in as root and issue the following command: 

cd /usr/src/kernels/2.6.9-34.EL-smp-i 686/include/linux 

wget http://nerdvittles.com/aah27/spinlock.h 

The rest of the procedure is as per earlier chapter Rebuilding Zaptel Driver 

E.5.2 As suggested by Rob Thomas (the FreePBX guru) 

Rob’s suggestion is to edit the file 


usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h 

and change 'rw_lock_t' on line 407 to 'rwlock_t' 

Once that is done, do the following: 

rebuild-zaptel 

That seems to be a lot easier. 

E.6 ZAPTEL CARD CONFIGURATION FOR AUSTRALIA 

Submitted by Stephen Gleeson – Technical Manager for Community Information Strategies 

Australia Inc (CISA). 

http://gleesos.wordpress.com/2006/02/08/voip-with-asterisk/ 

Stephen has attached samples of Zapata configuration files – with correct settings for 

Australia – loop start. 

zaptel.conf 

# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand 

edit 

# Zaptel Configuration File 

# 

# This file is parsed by the Zaptel Configurator, ztcfg 

# 

# It must be in the module loading order 

# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" 

# 

# Update for regional - ls - loopstart au - zone 

# 

fxols=1 

fxsls=2 

fxsls=3 

fxsls=4 

# Global data 

loadzone = au 

defaultzone = au 

zapata_addiotional.conf 

;;;;;;[230] 

signalling=fxo_ls 

record_out=Adhoc 

record_in=Adhoc 

mailbox=230@device 

echotraining=800 

echocancelwhenbridged=no 

echocancel=yes 


callprogress=no 

callerid=device 

busydetect=no 

busycount=7 


channel=>1 

zapata.conf 

; 

; Zapata telephony interface 

; 

; Configuration file 

[trunkgroups] 

[channels] 

language=en 


signalling=fxs_ls 

rxwink=300 

; Atlas seems to use long (250ms) winks 

; 

; Whether or not to do distinctive ring detection on FXO lines 

; 

;usedistinctiveringdetection=yes 

usecallerid=yes 

hidecallerid=no 

callwaiting=yes 

usecallingpres=yes 

callwaitingcallerid=yes 

threewaycalling=yes 

transfer=yes 

cancallforward=yes 

callreturn=yes 

echocancel=yes 

echocancelwhenbridged=no 

echotraining=800 

rxgain=0.0 

txgain=0.0 

group=0 

callgroup=1 

pickupgroup=1 

immediate=no 

;faxdetect=both 

faxdetect=incoming 

;faxdetect=outgoing 

;faxdetect=no 

;Include genzaptelconf configs 

#include zapata-auto.conf 

;Include AMP configs 

#include zapata_additional.conf 


E.7 RAID1, LVM (LOGICAL VOLUMES) – HOW TO 

(Submitted by a user: Rehan) 

Introduction 

According to Digium recommendations Asterisk boxes should be IDE based. This is 

counter to what one would expect. However if you are using Zaptel drivers then a fast 

hard disk or hardware RAID might ‘steal’ interrupts from the Zaptel drivers. Additionally 

many hardware RAID solutions set themselves up on the PCI bus with timings that can 

delay voice traffic to your voice cards. 

(See note 3 at the end if you want to use scsi or sata drives) 

For this particular application (A PBX) we are not too concerned about disk performance. 

Much of the software will run in memory on a 300Mb Ram machine (Trixbox default install 

consumes no more than 200Mb including the file cache). Fast hard discs improve 

performance in systems that are heavily dependant on a swap file. Linux will keep 

everything in memory + a reasonable file cache as much as possible. CPU performance 

and making sure you have sufficient memory to run the PBX are the key performance 

indicators of a healthy PBX. 

We wouldn’t want to put an Asterisk box in production without some kind of resiliency to 

survive continuous use over perhaps many years. To use an IDE disk based system we 

can use software RAID to provide some protection and to make sure the PBX stays up 

and running. The Raid subsystem can also email you if a drive fails. 

The following procedure describes setting up a software RAID during installation on all 

partitions of a Trixbox. This applies to a vanilla install from the Distribution CDROM. 

We will be using RAID and also Logical Volume Management. Logical Volume 

Management gives us the ability to dynamically resize drives and dynamically add disk 

space on a live system without having to delete or create new partitions. So if your 

requirements change in the future you can easily accommodate them. 

The issues: 

1. Implementing RAID and logical volumes post-install is complicated, dangerous and not 

recommended. 

2. RAID (hardware or software) and LVM are desirable features. IDE is still the cheapest 

storage platform. This is especially true of Asterisk installs as many PBX's use low-end 

hardware. 

3. The default install of Trixbox does not allow any options to install RAID or LVM 

4. The text mode Anaconda install does not allow manual configuration of Logical 

volumes however a kickstart file does. 

5. A kickstart install of RAID will always (!!!!) produce file system corruption if the software 

RAID is not allowed to complete its first sync before the first reboot. 

6. High end hardware RAID is still relatively expensive. Linux drivers are not always 

available or if available, not always stable. 

Point 5 will many times dissuade people from using software RAID and LVM. As long as 

we can accommodate the first sync then we will have no problems. 

Ok, lets get started: 


1) We’ll need to set-up the drives in a particular order: 

Hardware configuration: 

HD 1 connected to primary IDE as master. 

HD 2 connected to secondary IDE as master. 

CDROM connected to secondary IDE as slave. 

The hard disks don't need to be exactly the same size. Actually using this method the 

disks don't need to be the same size at all. I would recommend that the smallest disk be 

at least 5Gigs. The Raid will configure itself to the smallest disk and the larger disk will 

match up to the smaller disk’s partitions with any extra space left unused. 

Although I worked this for IDE drives it can be used on any type of disk subsystem - 

SCSI, sata etc with some modification for the drive labels. 

2) Extract/Edit the CDROM: 

Windows: 

I used UltraISO on windows to modify the CDROM. Open the ISO with UltraISO and copy 

the ks.cfg file somewhere. Remove the read only flag so you can edit it. It is very 

important you edit the file with an editor that knows how to do Linux linefeeds. I 

discovered and used Editpad pro for this. The lite (demo) version will suit the needs we 

have. If your editor is not 100% compatible with Linux linefeeds after you reboot you will 

get errors which have directory names with strange characters (/var/Trixbox_load^M/ or 

/var/Trixbox_load/R where the /rR is actually in the directory name of load not a separate 

directory) 

Linux: (incomplete) 

Extract the CDROM to a temporary Directory. Make the file ks.cfg as non-read only. Edit 

and create the files as below. Use the following command to rebuild the ISO. 

mkisofs -o modified-Trixbox.iso -b isolinux/isolinux.bin -c isolinux/boot.cat -no-emul-boot - 

boot-load-size 4 -boot-info-table 

3) Get the Installation configuration file as follows: 

We need to create a file that will tell the install program what to do. A copy of the existing 

file is located in the root of the CDROM. It is called ks.cfg. Make a copy of this file and call 

it ksRAIDlvm.cfg 

4) Edit the ksRAIDlvm.cfg as follows 

Delete the line where it says reboot. We don’t want it to automatically reboot at the end. 

This is a critical step! 

Where it says: 

#Partition clearing information 

clearpart --all –initlabel 

part /boot --fstype ext3 --size=100 

part / --fstype ext3 --size=1024 –grow 

part swap --size=256 --grow --maxsize=768 

Replace with 



clearpart --all –initlabel 

part raid.01 --size=100 --ondisk=hda –asprimary 

part raid.02 --size=100 --ondisk=hdc –asprimary 

part swap --size=256 --ondisk=hda 

part swap --size=256 --ondisk=hdc 

part raid.11 --size=1 --grow --ondisk=hda 

part raid.12 --size=1 --grow --ondisk=hdc 

raid /boot --level=1 --device=md0 raid.01 raid.02 

raid pv.01 --level=1 --device=md1 raid.11 raid.12 

volgroup vg01 pv.01 

logvol / --vgname=vg01 --size=10000 --name=root 

Very important: the –asprimary switch must only appear on the raid.01 and raid.02 

lines. 

This will give you a 100MB RAID 1 /boot partition, two 256Mb swap partitions (change the 

256 to a desirable size or replace with --recommended) and the rest of the disk will be a 

logical volume (over laying a RAID1 ) containing the / (or root) filesystem. All partitions will 

be formatted as Ext3 as normal. If you want to create other partitions to support other 

uses you can change the raid.11 and raid.12 lines to read --size=xxxx (in megabytes) and 

remove the –grow option. This will fix the size of the partition you create. After the 

installation completes you can then go ahead and create more RAID1 partitions and 

Logical volumes. 

Where it says (in the section starting - %post) 

mount /dev/hda /mnt/source 

change to 

mount /dev/md1 /mnt/source 

Add the following lines to the end of the %post% section. This will update the boot 

information to make both drives bootable. Remember to update the hdx numbers to 

match your drive configuration. 

/sbin/grub --batch /dev/null 

root (hd0,0) 

setup (hd0) 

root (hd1,0) 

setup (hd1) 

quit 

EOT 

Save the file. 

Add it to the root directory of the ISO and save the ISO under a different name. 

Burn the new ISO to a CD. Boot your soon to be asterisk machine from this CD and use 

the following command to start the install. 

linux ks=cdrom:/ksRAIDlvm.cfg 

The install process will continue as before however it will create the RAID as we have setup. 

Begin the install. Let it run through to the end. When you see the REBOOT screen STOP. 

Don't do anything here. We need to wait for the RAIDs to complete their first sync. 


A way to be sure that discs have been synced is to press Alt+F2 to get a new console 

and then execute the following command. 

watch cat /proc/mdstat 

Once you are sure the disks are in sync. Reboot. 

You are done! 

Checking and maintaining: 

(Some elements have been edited in collaboration with Sofoklis Sflomos who emailed me with 

some corrections*) 

Login as root using the password set-up during the earlier phase. 

You can check the state of the RAID by using the following command: 

cat /proc/mdstat 

This will tell you if any of the hard disks have crashed. 

 

Note by Sofoklis Sflomos: 

In case of hard disc crash, then: 

1. execute the command "cat /proc/mdstat" to indicate which disc has crashed (let's say 

that hdc has crashed) 

2. replace hdc 

3. restore the primary partition table with "dd if=/dev/hda of=/dev/hdc count=1 bs=512" 

4. restore the extended partition table with "sfdisk -d /dev/hda > bak-hda" 

5. and then "sfdsik /dev/hdc < bak-hda" 

6. mdadm -a /dev/md0 /dev/hdc1 

mdadm -a /dev/md1 /dev/hdc2 


You should have two raids called md0 md1 and md2 (if you are using 3 disks) or md0 and 

md1 (if you are using 2 disks). In square brackets for each it should say [2/2 (or 3/3)]. 

Your disks are ready and Trixbox is installed. The block sizes may be different if you are 

not using identical disks. 

If it doesn't say [2/2] depending on the number of hard disks you are using but the 

machine has restarted ok and it says [2/1] then one or both your raids have not synced 

properly or your disk is faulty (bad blocks). To manually sync the raids type some 

commands (for simplicity all the commands are included below. If a partition is already in 

a Synced RAID the command will fail with an error (in this case the error means 

success!). This you can ignore and move onto the next command): 

mdadm -a /dev/md0 /dev/hda1 







use 

watch cat /proc/mdstat 

to monitor the raid re-build. Once it is complete you can issue the 

reboot 

command and restart the machine. Check your raids again with 

cat /proc/mdstat 

They should be ok now. 

If the machine has not started ok (errors on the startup screen) then the chances are the 

RAID was not completely built before the restart. The best idea here is to re-install this 

time waiting longer before rebooting. It is difficult to know which files are not complete. 

The following is the full content of the ksRAIDlvm.cfg file as used by Sofoklis Sflomos. 

#CentOS trixbox Kickstart 

#modified by Sofoklis Sflomos ssflomos@cs.teiath.gr 

#System language 

lang en_US 

#Language modules to install 

langsupport en_US 

#System mouse 

mouse none 

#Use text mode install 

text 

#Install Red Hat Linux instead of upgrade 

install 

#Use CDROM installation media 

cdrom 

#System bootloader configuration 

bootloader --location=mbr 

#Clear the Master Boot Record 

zerombr yes 


clearpart --all --initlabel 

# Creates the BOOT partition that is 100mb in size 

part raid.01 --size=100 --ondisk=hda --asprimary 

part raid.02 --size=100 --ondisk=hdc --asprimary 

# Creates the SWAP partition (1024mb) 

part raid.11 --size=1024 --ondisk=hda 

part raid.12 --size=1024 --ondisk=hdc 


# Creates the root (/) partition that is able to grow to fill 



# Now create the raids using the above partitions 

raid /boot --level=1 --device=md0 --fstype=ext3 raid.01 raid.02 

raid swap --level=1 --device=md1 --fstype=swap raid.11 raid.12 

raid pv.01 --level=1 --device=md2 --fstype=ext3 raid.21 raid.22 

volgroup vg01 pv.01 

logvol / --vgname=vg01 --size 10000 --name=root 

#System authorization infomation 

auth --useshadow --enablemd5 

#Network information 

network --bootproto=dhcp --device=eth0 

#Firewall configuration 

firewall --disabled 

selinux --disabled 

#Do not configure XWindows 

skipx 

#Package install information 

%packages --resolvedeps 

alsa-utils 

audiofile-devel 

bind 

bison 

cpp 

curl 

curl-devel 

cvs 

dhcp 

flex 

gcc 

gcc-c++ 

gd 

glibc-devel 

glibc-headers 

glibc-kernheaders 

grub 

httpd 

kernel 

krb5-devel 

libc-client 

libcap 

libpcap 

libstdc++ 

libstdc++-devel 

libtermcap-devel 

libtiff 

libtiff-devel 

libxml2 

lm_sensors 

mc 

mkisofs 

mysql 


mysql-devel 

mysql-server 

ncurses-devel 

newt-devel 

net-snmp 

ntp 

openssl096b 

openssl-devel 

perl 

perl-libwww-perl 

perl-DateManip 

perl-DBD-MySQL 

perl-DBI 

perl-Net-DNS 

perl-suidperl 

perl-Time-HiRes 

perl-XML-Parser 

php 

php-gd 

php-imap 

php-mbstring 

php-mysql 

php-snmp 

postgresql-libs 

readline 

readline-devel 

sendmail-cf 

subversion 

system-config-date 

system-config-keyboard 

rpm-build 

slang-devel 

sox 

speex 

unixODBC 

vsftpd 

zlib-devel 

screen 

%post --nochroot 

mkdir /mnt/source 


mount /tmp/cdrom /mnt/source 

mkdir /mnt/sysimage/var/trixbox_load 

cd /mnt/sysimage/var/trixbox_load 

tar xvfz /mnt/source/trixbox.tar.gz 

#unmount CD-ROM 

umount /mnt/source 

# ------------------------------------------ 

# Post install 

# ------------------------------------------ 

%post 

echo "/var/trixbox_load/install_all.sh" >> /etc/rc3.d/S99local 

echo "NETWORKING=yes" > /etc/sysconfig/network 

echo "HOSTNAME=asterisk1.local" >> /etc/sysconfig/network 


echo "# Do not remove the following line, or various programs" > 

/etc/hosts 

echo "# that require network functionality will fail." >> 

/etc/hosts 

echo "127.0.0.1 

localhost" >> /etc/hosts 

echo "127.0.0.1 

asterisk1.local" >> /etc/hosts 

echo "driftfile /var/lib/ntp/drift" > /etc/ntp.conf 

echo "server 0.pool.ntp.org" >> /etc/ntp.conf 



echo "server 127.127.1.0" >> /etc/ntp.conf 

echo "fudge 127.127.1.0 stratum 10" >> /etc/ntp.conf 

cp -f /var/trixbox_load/splash.xpm.gz /boot/grub 

cp -f /boot/grub/grub.conf /boot/grub/grub.conf.old 

sed -e "s/hiddenmenu/#hiddenmenu/" /boot/grub/grub.conf.old > 

/boot/grub/grub.conf 

#make both disks bootable 


root (hd0,0) 

setup (hd0) 

root (hd1,0) 

setup (hd1) 

quit 

EOT 

eject 

Notes: 

1) In a VMWARE machine everything works fine. (To test this you need to create a 

second hard drive when creating the virtual machine then do the install booting either off 

the new CDROM or the new ISO). 

2) For anyone who wants to revert (or vert) an existing drive to Raid, here is a good link: 

http://linux-sxs.org/hardware/raid_for_idiots.html 

Just ignore the last bit and use grub instead (Lilo is too old!!!!). 

The following commands achieve the same ends: 

grub (enter) 

root (hdx0,0) 

setup (hdx0) 

root (hdx1,0) 

setup (hdx1) 

(ctrl-z to exit) 

3) This procedure will work for SCSI and SATA drives as well. Bearing in mind the 

caveats Digium have about using IDE many people have successfully installed RAID 1 on 

SCSI and SATA drives. To accomplish this replace the HDx with a suitable replacement 

(usually hd0 is replaced with sd0 and hd1 with sd1). 

If you have any issues booting from the RAID they can usually be fixed by booting from 

the Trixbox CD and starting the Rescue Environment and running some commands: 


Boot from the Trixbox CD. At the prompt type: 

Linux rescue 

Issue: 

grub (enter) 

root (hdx0,0) 

setup (hdx0) 

root (hdx1,0) 

setup (hdx1) 

(ctrl-z to exit) 

reboot 

We continue to use the textmode install because it reduces the chances of the install 

failing. For example if you have a Rage XL video card then using a graphical install, the 

format of the drives will fail (RAID or no RAID). In textmode logical volumes can only be 

created in the kickstart file. 

If your disk controller needs to have special drivers then the install must be started using 

the following command: 

Linux ks=cdrom:/ksRAIDlvm.cfg dd 

At some point it will ask you for a driver disk. This can usually be obtained from the 

manufacturers website. 


E.8 V2.2 TRIXBOX LOAD CONFIG TO INCLUDE IDE RAID 

The latest version of Trixbox includes a config file for sata raid (ks_sataraid.cfg) but not 

IDE raid. The following is the ks_ideraid.cfg for inclusion in Trixbox V2.2 (thanks to Rick 

Strangman for submitting this). 

#CentOS trixbox Kickstart 

#System language 

lang en_US 

#Language modules to install 

langsupport en_US 

#System mouse 

mouse none 

#Reboot after installation 

#Use text mode install 

text 

#Install Red Hat Linux instead of upgrade 

install 

#Use CDROM installation media 

cdrom 

#System bootloader configuration 

bootloader --location=mbr 

#Clear the Master Boot Record 

zerombr yes 


clearpart --all --initlabel 

# Creates the BOOT partition that is 100mb in size 

part raid.01 --size=100 --ondisk=hda --asprimary 

part raid.02 --size=100 --ondisk=hdc --asprimary 

# Creates the SWAP partition (1024mb) 

part raid.11 --size=1024 --ondisk=hda 

part raid.12 --size=1024 --ondisk=hdc 

# Creates the root (/) partition that is able to grow to fill 



# Now create the raids using the above partitions 

raid /boot --level=1 --device=md0 --fstype=ext3 raid.01 raid.02 

raid swap --level=1 --device=md1 --fstype=swap raid.11 raid.12 

raid / --level=1 --device=md2 --fstype=ext3 raid.21 raid.22 

#System authorization infomation 

auth --useshadow --enablemd5 

#Network information 

network --bootproto=dhcp --device=eth0 

#Firewall configuration 

firewall --disabled 

selinux --disabled 

#Do not configure XWindows 

skipx 


#Package install information 

%packages --resolvedeps 

alsa-utils 

audiofile-devel 

bind 

bison 

cpp 

curl 

curl-devel 

cvs 

dhcp 

flex 

gcc 

gcc-c++ 

gd 

glibc-devel 

glibc-headers 

glibc-kernheaders 

grub 

httpd 

kernel 

krb5-devel 

libc-client 

libcap 

libpcap 

libstdc++ 

libstdc++-devel 

libtermcap-devel 

libtiff 

libtiff-devel 

libxml2 

lm_sensors 

mc 

mkisofs 

mysql 

mysql-devel 

mysql-server 

ncurses-devel 

newt-devel 

net-snmp 

ntp 

openssl096b 

openssl-devel 

perl 

perl-libwww-perl 

perl-DateManip 

perl-DBD-MySQL 

perl-DBI 

perl-Net-DNS 

perl-suidperl 

perl-Time-HiRes 

perl-XML-Parser 

php 

php-gd 

php-imap 

php-mbstring 

php-mysql 

php-snmp 

postgresql-libs 

readline 

readline-devel 

sendmail-cf 

subversion 

system-config-date 

system-config-keyboard 

rpm-build 


slang-devel 

sox 

speex 

unixODBC 

vsftpd 

zlib-devel 

%post --nochroot 

mkdir /mnt/source 


mount /tmp/cdrom /mnt/source 

mkdir /mnt/sysimage/var/trixbox_load 

cd /mnt/sysimage/var/trixbox_load 

tar xvfz /mnt/source/trixbox.tar.gz 

#unmount CD-ROM 

umount /mnt/source 

# ------------------------------------------ 

# Post install 

# ------------------------------------------ 

%post 

echo "/var/trixbox_load/install_all.sh" >> /etc/rc3.d/S99local 

echo "NETWORKING=yes" > /etc/sysconfig/network 

echo "HOSTNAME=asterisk1.local" >> /etc/sysconfig/network 

echo "# Do not remove the following line, or various programs" > 

/etc/hosts 

echo "# that require network functionality will fail." >> /etc/hosts 

echo "127.0.0.1 

localhost" >> /etc/hosts 

echo "127.0.0.1 

asterisk1.local" >> /etc/hosts 

echo "driftfile /var/lib/ntp/drift" > /etc/ntp.conf 




echo "server 127.127.1.0" >> /etc/ntp.conf 

echo "fudge 127.127.1.0 stratum 10" >> /etc/ntp.conf 

cp -f /var/trixbox_load/splash.xpm.gz /boot/grub 

cp -f /boot/grub/grub.conf /boot/grub/grub.conf.old 

sed -e "s/hiddenmenu/#hiddenmenu/" /boot/grub/grub.conf.old > 

/boot/grub/grub.conf 

# make both disks bootable 


device (hd0) /dev/hdc 

root (hd0,0) 

setup (hd0) 

device (hd0) /dev/hda 

root (hd0,0) 

setup (hd0) 

quit 

EOT 

eject 


E.9 TEST BENCH ANYONE 

(Contribution from zforum69) 

Issue: I don't have any room for test machines, let alone spare TDM400 cards, to try out 

new releases of trixbox or other Asterisk distros like elastix. I cannot have my main stable 

trixbox machine down for any length of time. 

Solution: Install each different version on a separate partition, and change back to the 

production partition when you are done testing. 

How: 

When booting a trixbox install CD, normally you just press enter after the boot prompt, 

where it will promptly delete everything on the disk and install trixbox to occupy the whole 

disk. Instead of pressing enter, type advanced where you can control how the partitions 

are created and where trixbox will be installed. During the install it is a matter of following 

your nose until it gets to the part about disk partitioning, which you will choose to do 

manually rather take what ever the install is offering. 

If this going to be the first install on this disk delete all existing partitions if they exist. 

Create a new partition and set the mount point to "/" so it will install it into that partition. 

Set the size at say 20GB. Create a swap partition (2GB). Install the GRUB boot loader 

into the “Master Boot record”. 

If this second or subsequent Asterisk install on this disk, just create a new primary 

partition, again with the mount point set to "/" so it will install it into the newly created 

partition. You don't need to install another swap partition as there should already be one 

there from a previous installation. Install the GRUB boot loader into the "First sector of 

boot partition". 

Only the GRUB boot loader from the first install in the MBR will ever be used. The 

“grub.conf” generated by each other install is used to add to the grub.conf in the first 

install to get a menu of installs you can boot to. 

Boot the first install and mount the other partitions to get to look at their “grub.conf” files. 

Using those files edit the “grub.conf” in the first install to add the additional menu items for 

all your other installs. Now when ever you boot up you can interrupt the GRUB boot 

loader and select which ever install you want to boot to next. To do it remotely without a 

remote console you need to modify the “default=0” line in the grub.conf. That value 

determines which of the menu items the default is. By changing that value the next boot 

up and every subsequent one will cause GRUB to default to the new menu item and boot 

that. To change it back, mount the first partition, and change the default line back to "0" in 

the “grub.conf” file. 


APPENDIX F (PUBLICATIONS AND REFERENCES) 

F.1 DUNDI – A GUIDE TO A SIMPLE SETUP 

You will find a guide to a simple DUNDi set-up at the GoldfishNet Blog: 

http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/ 

F.2 THE FUTURE OF TELEPHONY BY O’REILLY PUBLISHING 

You may download the pdf version of this publication here: 

http://voipspeak.net/images/stories/orielly/AsteriskTFOT.zip 

O’Reilly has licensed the book under the creative commons license and as such, you are 

free to download and distribute it. 

F.3 OTHER ASTERISK FORUMS, TUTORIALS AND WIKIS 

One of my favourite site is http://www.asteriskconnection.com.au/ 

Others that I refer to are: 

http://asteriskathome.sourceforge.net/ 

http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki 

http://asteriskathome.sourceforge.net/faq.html 

http://sourceforge.net/forum/group_id=123387 

http://amp.coalescentsystems.ca/ 

http://sourceforge.net/docman/group_id=121515 

The following are two tutorial sites that I recommend everyone who is serious about 

Trixbox and Asterisk generally. 

http://mundy.org/blog/index.phpp=81 

http://www.asteriskguru.com/tutorials/ 

General VOIP related: 

http://www.freewebs.com/voipformula/VoIP-HOWTO.html 

F.4 VOICE OVER IP – PER CALL BANDWIDTH CONSUMPTION 

The following is a reference from Cisco Systems that’s worth a look. 

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml 


NOTES 

A2BILLING 

A2Billing is a calling card platform to manage calling card users’ accounts. To log on to 

A2Billing through Web Interface, connect to asterisk.ip.address/a2billing e.g. 

192.168.0.101/a2billing 

Login as admin using the default password of mypassword. 

A2Billing is not being covered in this guide, as it is a whole documentation exercise on its 

own. Those interested in A2Billing may have to refer to other source for instructions 

available on the net such as the following link: 

http://sourceforge.net/forum/forum.phpthread_id=1398290&forum_id=420324 

http://www.voip-info.org/wiki/view/Asterisk+billing 

DELETE CALL RECORDS FROM CDR 

There will be time when you need to delete the CDR records from your call details record. 

One example is, those calls you made white testing will need to be deleted before going 

live where you will need clean record. 

• These records are being kept in the SQL database and to delete them, do the 

following: 

• From Tools Menu (or from System menu for Trixbox 2.2) 

• select phpmyadmin 

• In the Database dropdown field, (top left), select the asteriskcdrdb. 

Select the 'cdr' table in the tick box 

• At the bottom of the list of table you will find a dropdown field “With Selected”, 

select EMPTY 

• You will be asks to 'confirm truncate table cdr', 

• If you click Yes at this point the content of the table will be erased. 

 

MySQL Manager – “tbm-phpmyadmin.noarch” must be installed. This can be done 

through Packages. It is not installed by default. Without this you will not be able to 

use phpadmin from the GUI. 

Alternatively, you can do it manually; 

Login to the shell (SSH) as root. You can do this using Putty. 

Enter these commands manually, each on a new line. 

mysql –p 

Enter password: passw0rd 

use asteriskcdrdb; 

delete from cdr; 

exit; 

Your CDR will now be empty. 


EXTENSION NUMBERS TO AVOID USING 

Unless you are prepared to edit and change some codes, its best to avoid the following 

extension numbers: 

200 - Park Notify 

300-399 - Reserved for speed dial 

666 - Reserved for FAX testing 

70-79 - Reserved for calls on hold 

700-799 - Reserved for calls on hold 

7777 - Reserved extension for incoming calls simulation 

SIP_NAT.CONF 

To those who use this configuration (not all do), I have an issue with externip = . I wasn’t able to accept incoming calls although SIP was registered. Changing to 

externip= if you have a fixed IP or externhost= if 

you don’t have a fixed IP and use Dynamic DNS service, solved the issue or do the 

alternative as I have outlined below. 

This issue may not be apparent at first as Asterisk will show that it is registered and you 

will still be able to make a call out (except calls to another Asterisk that have this property 

set). Another telltale sign is, you may not hear the ring tone when you are calling a party 

or you may have audio problem. 

This is caused by my /etc/hosts file where I have an entry (in red) 

127.0.0.1 localhost 

127.0.0.1 pcnovation.homelinux.org asterisk1.local 

SipBroker will resolve pcnovation.homelinux.org as 127.0.0.1 which is what it is suppose 

to be as it is being defined as such by the offending line. In actual fact 

pcnovation.homelinux.org should be resolved to the external IP address. However I need 

that entry for my VoiceMail e-mail notification since most email server will not accept 

email from unknown source – grief time. 

Taking my DNS name out solved the issue, but I cannot send e-mail notification of my 

voicemail. To fix this problem (a kludge) I substituted pcnovation.homelinux.org with 

another valid domain name different to my DynDNS domain name such as yahoo.com 

thus: 

127.0.0.1 yahoo.com asterisk1.local 

and problem solved… for now until something else pops up ☺ 

HOW TO RESTRICT OUTGOING CALLS – MANUAL METHOD 

This requirement often manifests itself in an office environment where management 

restricts staff from making International, long distance, or even local calls. In many 

organizations, staffs are confined to making internal calls only. 

There are 2 ways that this can be handled. One is by using the Custom Context module 

or by creating the necessary context manually. 

Here, we will create the procedure manually. 


It is somewhat complicated, but I will try to explain it here and provide an example. 

1. Set up all your trunks and outbound routes as normal. At this stage, users can make 

unrestricted calls. This is the normal way of doing it with freePBX. You probably 

would have already done this. 

2. Through Config Edit of Trixbox, open extensions_additional.conf and locate the 

[outbound-allroutes] section of the file. You will notice something like my example 

below. 

[outbound-allroutes] 

include => outbound-allroutes-custom 

include => outrt-001-PSTN 

include => outrt-002-SPA3K 

include => outrt-003-Oztell 

include => outrt-004-Pennytel 

include => outrt-005-Domestic 

include => outrt-006-MobileAust 

include => outrt-007-International 

include => outrt-008-12Number 

include => outrt-009-13N1800Numbers 

include => outrt-010-e164 

include => outrt-011-SingaporeLink 

include => outrt-012-KualaLumpurLink 

include => outrt-013-MalaccaLink 

exten => foo,1,Noop(bar) 

; end of [outbound-allroutes] 

Highlight them, copy and paste in a notepad text file somewhere and will get back to 

this a little later. 

Alternatively you can paste it directly to extensions_custom.conf. Do not do this if 

Trixbox is active as it may cause some problem while people using it. 

3. Through Config Edit of Trixbox, open extensions_custom.conf and create a 

section like my example below. You can get the codes from [from-internaladditional] 

of extensions_additional.conf. So will have to do some juggling 

because you can’t cut and paste this from another .conf file without having to close 

one (you can cut and paste from here if you like). 

[from-restricted] 

; 

; These are all the applications that you will require 

; 

include => app-cf-busy-off 

include => app-cf-busy-off-any 

include => app-cf-busy-on 

include => app-cf-off 

include => app-cf-off-any 

include => app-cf-on 

include => app-cf-unavailable-off 

include => app-cf-unavailable-on 

include => app-calltrace 

include => app-callwaiting-cwoff 

include => app-callwaiting-cwon 

include => app-dialvm 

include => app-directory 

include => app-dnd-off 

include => app-dnd-on 

include => app-echo-test 


include => app-recordings 

include => app-speakextennum 

include => app-speakingclock 

include => app-userlogonoff 

include => app-zapbarge 

include => app-vmmain 

include => ext-group 

include => ext-fax 

include => ext-meetme 

include => ext-findmefollow 

include => ext-paging 

include => ext-queues 

include => ext-test 

include => ext-local 

include => parkedcalls 

; 

; #### OutBound Routes ######## 

; # Below are all restricted routes # 

; ########################### 

; 

Now copy your [outbound-allroutes] that was saved to the text file directly underneath 

the last line of the above context. Remember to delete the [outbound-allroutes] label. 

Unfortunately you can’t cut and paste my example because they are my routes NOT 

yours. 

Your final handiwork will look something like the example below: 

[from-restricted] 

; 

; These are all the applications that you will require 

; 

include => app-cf-busy-off 

include => app-cf-busy-off-any 

include => app-cf-busy-on 

include => app-cf-off 

include => app-cf-off-any 

include => app-cf-on 

include => app-cf-unavailable-off 

include => app-cf-unavailable-on 

include => app-calltrace 

include => app-callwaiting-cwoff 

include => app-callwaiting-cwon 

include => app-dialvm 

include => app-directory 

include => app-dnd-off 

include => app-dnd-on 

include => app-echo-test 

include => app-recordings 

include => app-speakextennum 

include => app-speakingclock 

include => app-userlogonoff 

include => app-zapbarge 

include => app-vmmain 

include => ext-group 

include => ext-fax 

include => ext-meetme 

include => ext-findmefollow 

include => ext-paging 

include => ext-queues 

include => ext-test 


include => ext-local 

include => parkedcalls 

; 

; #### OutBound Routes ######## 

; # Below are all restricted routes # 

; ########################### 

; 

; new outbound-restricted-routes 

include => outbound-allroutes-custom 

;include => outrt-001-PSTN 

include => outrt-002-SPA3K 

include => outrt-003-Oztell 

include => outrt-004-Pennytel 

include => outrt-005-Domestic 

include => outrt-006-MobileAust 

include => outrt-007-International 

include => outrt-008-12Number 

include => outrt-009-13N1800Numbers 

include => outrt-010-e164 

include => outrt-011-SingaporeLink 

include => outrt-012-KualaLumpurLink 

include => outrt-013-MalaccaLink 

exten => foo,1,Noop(bar) 

; 

; end of outbound-restricted-routes 

Now add the following codes underneath that to finish it off. 

; 

exten => h,1,Hangup 

exten => s,1,Macro(hangupcall) 

exten => h,1,Macro(hangupcall) 

4. You may now comment out the outbound routes you do not want the restricted 

extensions to use. 

E.g. If you comment out include => outrt-001-PSTN, the restricted extension 

cannot use PSTN. 

5. After you have done that, you need to go back to FreePBX and edit the extension that 

you want to restrict by change the context to from-restricted instead of from-internal. 

Now go enjoy yourself with it. 


BUGS REPORTS 

Software is not software if it is release without any bug. It just goes against nature’s logic, 

not to mention our Mr. Murphy ☺ 

ASTERISK RE-LOAD ISSUE 

After making changes to Trunks, IVR or DISA etc, the changes do not translate to the 

system. This is due to an inherent bug in the Asterisk binary in Trixbox 1.2. 

If this happens, run the following at the command prompt to force the reload: 

asterisk -rx restart now 

If the problem is related to Asterisk, this should get you going otherwise, it may be some 

other gremlin that has gotten into the system. 

ARE YOU STILL HAVING PROBLEM 

If you are still having problem installing Trixbox 1.2, it probably is time for you to do some 

manual configuration as described in Rob Thomas’ blog. 

Here is the link and as I have personally done it this way, I can assure you that it will 

solve your problem – I hope ☺ 

http://www.freepbx.org/2006/09/28/un-Trixbox-your-Trixbox/#more-7 

HANGS ON SHUTDOWN 

Using X100P card caused Trixbox to hang on shutdown in Trixbox 2.x. This happened to 

me on 2 different make X100P. This does not happen with TDM400 card. 

To avoid this hanging anomaly amportal stop must be executed before a reboot or 

shutdown. 

The following is the hack to stop the panic from occurring: 

Log in as root: 

Go to the /etc/rc.d directory. 

cd /etc/rc.d 

Rename all of the K92zaptel files to K95zaptel so that they are invoked after the 

K94asterisk scripts: 

mv rc0.d/K92zaptel rc0.d/K95zaptel 





shutdown -r now 

The system will now shutdown or reboot without a panic. 


ZAPTEL NOT DETECTED ON TRIXBOX 2.2.4 

Trixbox 2.2.4 seems to have a problem detecting Zaptel cards e.g. TDM400 from fresh 

install. To remedy this problem, it may be necessary to run the following command: 

yum clean all 

yum install zaptel* 

ALL PHONES UNUSABLE IF INTERNET CONNECTION IS LOST. 

This only happens if users are using SIP Trunks. Luckyly, those using IAX trunks only 

are spared from this issue. 

Some say it is not a bug but to me if users have to make modification to get all the 

phones to work locally and accept PSTN calls, something is terribly wrong and therefore it 

is a bug and I think it is a fundamental flaw in the design of Asterisk. 

Asterisk is assuming that when a call is being made, the call is going to be made via a 

trunk IF a sip trunk is recorded and enabled. Therefore Asterisk will scan for SIP trunks 

availability. Once Asterisk finds a trunk it will start sip. If it does not find an available trunk 

it will give up and start sip then you can make a call. However if you have more than one 

SIP trunks, it will scan for all enabled SIP trunks until the verification times out. If you also 

happen to have 10 SIP trunks, all 10 will be verified first before you can even use your 

phones. 

In normal cases where internet connection is available, this will only take a couple of 

seconds to complete, but if internet connection is out, the verification process will take for 

ever till it just die away and all your extensions rendered useless. 

If no SIP trunk is recorded or enabled at all, then it will start sip immediately and you can 

make a call. This issue only appears for those using SIP trunks, especially multiple SIP 

trunks. 

This is not a major issue if you only have 1 sip trunk because when Asterisk failed on that 

trunk, it will give up and start sip. The only thing you notice will be a slight delay between 

your dialling and the other phone ringing. Normally it is just a slight irritation - nothing 

more. 

The problem starts when you have multiple SIP trunks. If you have 2 SIP trunks, the delay 

before Asterisk gives up is a little longer and you can still dial internal number after a long 

delay. In many cases the delay is enough for users to think that nothing is happening, but 

if you keep waiting it will dial. 

However if you have 3 or more trunks, the delay becomes so long for Asterisk to cycle 

through all your SIP trunks and the phone system becomes unusable, just gave up and 

dies. 

This is where the fundamental flaw is. 

Asterisk should dial the number without scanning for SIP trunk (even if you have SIP 

trunks) unless the number dialed is part of an outbound route that requires SIP trunk. 

IAX is spared this hassle why not SIP 

I tested this by adding 1 sip trunk then 2 sip trunks etc. The delay becomes progressively 

longer and longer the more sip trunks I added.... which brought me to the above 

conclusion. 


Question is; how do we get around this There is a way but it is a kludge. It works and 

you don’t have to fiddle with codes or create a local DNS or BIND and what not that the 

normal digger would not know how. 

This is a project you should try. 

Hint: Create 2 boxes. Link both boxes using SIP. Make the other box a SIP trunk of your 

live box. Make sure that this SIP trunk appears as the first trunk in your 

sip_additional.conf. Comment out externhost and exterip from sip.conf or sip_nat.conf.. 


CHEAT SHEET 

amportal restart 

asterisk -rvvvv 

Disable default voicemail 

message 

Enable Call waiting by default 

grep "AUTO FXO" /var/log/messages 

/etc/init.d/ntpd stop 

ntpdate ntp.netspace.net.au 

/etc/init.d/ntpd start 

rpm -qa | grep asterisk 

rpm -qa | grep zaptel-modules 

service asterisk restart 

service network restart 

Set System time on system clock 

core show features 

core show translation 

sip reload 

To restart Trixbox after configuration change without 

rebooting. However certain changes will need 

rebooting. 

To get to asterisk CLI 

This works in custom context e.g. 

• Voicemail(s2000@default) - will not play 

default message 

• Voicemail(su2000@default) - will not play 

default message and instead play your 

unavailable custom message. 

Edit ENABLECW=yes in amportal.conf 

Tells you what mode your TDM400 is running on. 

To set the NTP time manually from the command line. 

Tells you what version of Aterisk is installed. 

Tells you what Zaptel modules you have installed 

To restart asterisk if required when something goes 

wrong 

To restart network service if connection goes down and 

did not come up again even though internet has been 

restored. 

To set the system clock under Linux, you need touse 

the “date” command. Example: To set the current time 

and date to May 12, 2007:10:15.30 seconds pm, type 

``date 051222152007.30' (The time,in bold, is in 24 

hour notation). To see what the current local time is, 

run “date” with no arguments. 

This command run on the Asterisk CLI to tell you extra 

featuress in use 

This command run on the Asterisk CLI to tell you 

codecs in use. 

To reload SIP. This command is to be run from 

Asterisk CLI 

SOME LINUX COMMANDS IN COMMON USE 

chown 

To change file ownership to the new owner 


BIBLIOGRAPHY 

The real people behind this guide (Real Names used or aliases if real names unknown): I 

have also made references to a number of other online publications and forums whenever 

I need confirmations and further clarifications. Some of the information in this document 

were sourced from these people and places. 

1 Andrew Gillis The man who started it all – The Mahaguru 

2 Asterisk Guru http://www.asteriskguru.com 

3 Brian Advise tips and tricks- Voice for Asterisk 

4 Bob Fryer A friend from a long way back. From the days I learned to 

use a PC and according to Bob, I still can’t use it. 

5 Colin Swan Major assistance, advice, and basically a place to go when 

things go wrong. 

6 Graham Foote Always giving me tips and corrections 

7 Greg Hind Advise, Tips and tricks 

8 Jack Zimmermann Helping me with e164 

9 Jeffrey Borg Advise, Tips and tricks 

10 lusyn.com X100P Patch for UK Caller ID http://www.lusyn.com 

11 Mark Brooker Another of my source for help when things go wrong 

12 Matt (daggo) Advise, Tips and tricks 

13 Neilmc Advise, Tips and tricks 

14 Nathan Poyner Assistance with Scripts and also proofing the documents 

15 Openvoice Source for Australian Voice 

16 Peter Quodling Testbed and resident cynic 

17 Rehan Help with Raid configuration. 

18 Rob Thomas freePBX Developer and Super Hero – without which there 

will be no Trixbox 

19 Sammy/sz Taken time to point out some of my mistakes 

20 Sean Mahon Solution for Billion 7402 BL 

21 Shaun (Ewing) Advise, Tips and tricks on codecs and dial plan etc 

22 Sofoklis Sflomos Changes and assistance with Raid configuration 

23 Stefan Keller-Tuberg Helping me out with Cron Jobs and Scripts 

24 SteveM Advise, Tips and tricks on Webmin 

25 Steven D Help with detecting some error and proofing the document 

26 Thunderbird1/Chippy Giving me clues to some obscure processes. It is Chippy who 

solved the Fax problem that has been bugging us all. 

27 voip-info.org A site I often refer to when I require further clarification. 

28 VoipShop Source for Australian voice and others 

29 Voxilla.com http://voxilla.com 

30 Ward Mundy My hero http://mundy.org my place of salvation ☺. 

31 The PiaF Dev Team For giving us PiaF (Thanks Ward) 


CHANGES MADE SINCE LAST UPDATE 

Included some changes due to Asterisk 1.4

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