ElastixEasy
By Haamed Kouhfallah
PDF Guide by Haamed, 23 Jan 2012
ELASTIX
EASY
[Haamed Kouhfallah]
ElastixEasy is a reference book for who interested in Asterisk and Elastix
and typically the voip technology. I‘d like to dedicate this book to all
children suffering from pediatric cancer. This book is free, though, if you
find it useful; you can donate to your local pediatric cancer foundations.
Thanks in advanced.
ELastix Easy
For ELastix 2.x and Freepbx 2.x
http://www.voip-iran.com
by
Haamed Kouhfallah
‫حامدكوهفالح‬
Email: k.haamed@gmail.com
Last Update: 18 Jan 2012
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ElastixEasy by Haamed Kouhfallah
About the author [Haamed Kouhfallah]:
His experience in the field of VoIP includes:
 Elastix Engineering (ECE) from Palosanto Group;
 Community member of Elastix website on June 2009 and among
the top ten members of Elastix.org in 2009;
 Community member of Elastix website on December 2010 and
among the top ten members of Elastix.org in 2010;
 Produce Vaak telephone system that is the translation of Elastix
system into Persian, he also provide package of Persian converter
of Elastix per each copy provided from Elastix website;
 Author of Elastix in Persian as the only Persian reference and Free
PBX which is introduced as reference book in www.elastix.org ;
 conducting training courses of Elastix and VOIP in Iran;
 Official trainer of Elastix Engineering (ECE & ECT) in Iran.
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ElastixEasy by Haamed Kouhfallah
1.
INTRODUCTION
2.
VOIP HARDWARES
2.1
2.2
2.3
3.
INSTALLATION
3.1
3.2
3.3
3.4
3.5
3.6
4.
Prerequisites
Installing ISO
Network configuration
The default usernames
Accessing to the graphical enviroment
changing the admin password
USEFUL PROGRAMS
4.1
4.2
5.
Webmin
WinSCP
14
14
17
19
20
20
20
25
27
28
28
29
29
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TELEPHONY CONFIGURATION (PBX)
33
5.1
General setting:
5.2
Extensions
5.3
setup softphones:
5.4
follow me
5.5
What is dial pattern?
What is trunk?
5.6
5.7
Outbound Routers:
5.8
inbound routs:
5.9
Time Conditions
5.10
system recording
5.11
final destinations
5.12
digital receptionists (IVR)
5.13
Ring Group
5.14
Queue
5.14.1
What is queue
5.14.2
Kind of queue
5.14.3
Making queue
5.15
Announcement
5.16
Call back
5.17
Accessing to dial tone from outside (DISA)
33
38
43
50
52
53
55
57
59
62
62
63
66
69
69
69
69
76
78
81
6.
FAX SERVER
6.1
6.2
6.3
6.4
3
Access methods and using the services of VoIP
Computer
Telephony card
6
Making a IAX Extension
Visual Fax
View the Faxes
Programs of sending fax
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84
84
85
85
7.
CALLCENTER MODULE
7.1
7.2
7.3
7.4
7.5
7.6
7.6.1
7.6.2
7.7
7.8
7.9
7.10
8.
9.
introduction
Concept of call center
installing call center module
use of call center
making agent and its configuration
ingoing calls
Queues
Clients
Breaks
Forms
outgoing calls
Console environment:
92
92
92
93
93
95
96
96
97
98
100
104
CUSTOM CONTEXT MODULE
105
INSTANT MESSAGING WITH OPENFIRE
111
9.1
9.2
10.
Installing Openfire
Installing SPARK
ROOMX MODULE
111
126
129
10. 1
10. 2
Configuration
Models
129
131
3 .01
Mini-Bar
Checking
132
133
Room List
Group List.
Checkout
Billing Report
Booking
134
135
135
137
137
10. 4
5 .01
10. 6
10. 7
10. 8
10. 9
01 .01 Home
11.
11.1
11.2
11.3
11.4
11.5
11.6
11.7
11.8
12.
12.1
12.2
12.3
4
92
139
ASTERISK AGI PROGRAMMING
141
What is AGI?
Calling an AGI Script from the Dialplan
AGI, EAGI, DeadAGI and FastAGI
AGI scripting frameworks
The ten rules of AGI development
AGI Commands
simple PHPAGI example
Interact with MySQL inside PHP-AGI
141
141
142
143
144
148
151
153
TIPS AND TRICKS
connecting 2 servers of Elastix together
Video call on Elastix
Limiting the conversation time
ElastixEasy by Haamed Kouhfallah
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155
166
168
12.4
12.5
12.6
12.7
12.8
12.9
12.10
12.11
13.
3.1
3.2
3.3
3.4
3.5
5
put your Asterisk server behind NAT
Installing Codecs of g729 & g723
Asterisk command-line interface (CLI)
Asterisk feature codes
Reading asterisk log files
Asterisk integration with Cisco Call Manager
Customized chanspy
Tips and tricks of reading log files
TROUBLESHOOTING AND MAINTENANCE
Heavy Asterisk Log
How to tackle Jitter issue
Echo on POTS lines
Essential Linux commands
job scheduling in linux using crontab
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171
177
178
179
183
184
185
185
187
188
190
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1. Introduction
1.1
About Asterisk
Asterisk is open source software used for unified communications which
was first created in 1999 by Mark Spencer, the managing director of
Digium Company, based on GNU/GPL license. Asterisk is a step towards
generalizing various communication methods based on computer & local
networks for audio-visual telecommunications & related applications
such as IM, Call/Contact Centre & etc.
As IP based computer network communications grows, Asterisk faces
ever increasing success. Being free for all when compared to the
enormous prices of the current brands in the market, in addition to
having various potential capabilities with adequate quality, standard
protocols, not being limited/depended to a particular brand of software
or hardware, easy to install & operate, the sheer size of its third party
developer community & most importantly, unified voice (whether voice
or telephone), visual & data services have made Asterisk as a soft switch
to become one the effective & dynamic components of the next
generation of communication.
Asterisk is based on C programming language and is loaded in various
operating systems such as Linux NetBSD, UNIX, Solaris, Mac OSX,
FreeBSD, and OpenBSD. In addition, other versions of Asterisk can be
installed in windows platform. Although by using computers, common
servers and calculating the power of system (CPU/RAM) based on the
number of users, Asterisk services can become operational, but the
popularity and variety of its services prompts many manufacturers to
use combined platform of Linux and Asterisk to make unified
communications equipment in different scales. From very simple,
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efficient and cheap equipment in SOHO & SMB scale to complex designs
with large number of users in Enterprise environments, production of
such tools is very easy and simple because their software is available
and it is enough to facilitate the operation of system by designing
appropriate interface and web based. In more complex samples,
because of open source of Linux and Asterisk, changing the source of
software can be possible for better performance.
As multipurpose software which is based on information networks the
best thing to do is to designing a network (QoS, Redundancy, Traffic
Management& planning) and using its hardware appropriately in SMB &
Enterprise environments. Thus, Asterisk should have these requirements
whether it is used in simple application like phone center (IPBX), more
complex like video conference and call/contact center or in unification
with software such as office automation, ERP and etc.
Contrary to many, not only Asterisk and basically soft switch idea, audio
and video communications based on network application is not in conflict
with traditional view of telecommunications but also it has complement
and developer role. Although Asterisk is popular, its communication
based on computer network (Video Conferencing, IP Technology, VoIP,
etc...) is cheap and extended, but justification with traditional structure,
generally TDM, is not forgotten in Asterisk and more importantly
communication media has no effect in its operation. Set up Asterisk
based on IP Based equipment is easier and cheaper but justification with
older technologies should be considered as well. Security and reliability
of operation in soft switch systems and Asterisk in compare with
traditional communicative systems is a reason of conflict between soft
switch and traditional ideas. These two articles should be discussed
separately but at the end solution of an Asterisk system for security and
operation reliability is shortly expressed.
Encryption of communication is the best way which line tapping and
having fast computers cannot decode it. In addition to common ways,
proprietary protocols can be made in encryption of communication. This
is possible because of capability of Linux operating system and for
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providing the communications security between systems components
based on Asterisk both common encryption and propriety protocols can
be used. Beside this Linux is an appropriate firewall that can guarantee
the security coefficient of accessing the Asterisk services to the high
level beside other firewalls.
Most of the typical features of Asterisk system, which is installed in Linux
platform, are actually taken from Linux operating systems. High power
capabilities such as Clustering & HA (High Availability) of Linux
guarantee the operation reliability of soft switch system based on
Asterisk. Besides, hardware redundancy like power supply with
redundancy of computer network in links, equipment, protocols and etc…
cause that Asterisk be in the same level with TDM Based systems. So
Asterisk is a way toward presenting next generation services in divers
scales (Enterprise,SMB,SOHO). A way which leads to unified
communications, innovation and simply providing extensive range of
audio and video services and also Fixed Mobile Convergence Enterprise
in organizational interactions.
1.2
About Elastix
Elastix is a collection of best open source programs and tools which are
combined together and finally create a comprehensive IP PBX. It is
designed properly and gives you a PBX system that can compete with
others, not only because of PBX part but also because it is capable of
creating a powerful system with other products and programs.
The most important parts of Elastix:
 Asterisk: as the core PBX (Digium‘s most well-known product)
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 VTigerCRM and SugarCRM: as a communication system with
customers
 A2Billing: program to pay bills of Asterisk
 Flash operator panel: operator console which is like monitor
display.
 Hylafax: a software fax system
 Openfire: a server with dialogue system, sending text and
telephone network
 Conferencing: is an controlling devise
 freeBPX:an application tool for Elastix
 A report system: part of Elastix that provide CD report
 OSLEC: it is a software that remove echo sound
 Postfix: a popular mail server
 Round cube webmail: an interface for using web based mail
services
 CentOS: it is a version of Linux, Redhat with free support, and one
copy of Centos will be released by each copy of Redhat. They both
were supported and produced by different companies and in many
cases big and small companies uses these for manufacturing their
products. Elastix producers compile a web interface to access the
programs which seems to be complete. Also Elastix company
provide a software for reporting, diagnosing the hardware, network
setting, module of updating software, backup module, managing
users and other modules.
1.3
About freepbx
When Asterisk suddenly became popular, many company started to
invest on it. Asterisk does not have graphical environment for easy set
up so companies started to design a graphical environment for that.
Some of them were free and open and others commercial. One of the
graphical environments is freepbx which is popular because of providing
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many applications for users. Some companies that produce ISO for
Asterisk, use this program for managing and setting up the Asterisk such
as Trixbox, Elastix, Asterisk now…
The official website of this program is:
http://www.freepbx.org
1.4
Common terms/phrases in VoIP
There are some common phrases in VoIP technology
Definition
10
abbreviation
adaptor of analogue
telephone to IP based
telephone
Analogue Telephone
Adapter
ATA
Port of RJ11 is like a
telephone and
equipment attached
to it should be
capable of receiving
alarm signal
answering and ending
a call
Foreign Exchange
Office
FXO
Open line which is
attached to RJ11 can
provide an open line
and power of FXS in
an analogue line to a
telephone
Foreign Exchange
Station
FXS
Network based on
honeycomb model
Global System for
Mobile
GSM
ElastixEasy by Haamed Kouhfallah
11
which is the most
common standard for
mobile equipment
Communication
Proprietary protocols
of Asterisk with
RFC5456
Inter Asterisk
eXchange
protocols(version 2)
LAX (IAX2)
Internet
standardization
committee
Internet Engineering
Task Force
IETF
Committee of the
international
telecommunication
International
Telecommunications
union
ITU
It refers to
automated telephone
answering system
Interactive Voice
Response
IVR
It is an interface
between local
network and an
internet. NAT allows a
device in internet to
work like a router
Network Address
Translator
NAT
It refers to local
telephone centers.
The main task of PBX
is exposure between
one or some
telephone line and
some of users and
also dividing bilateral
contacts
Private (Automated)
Branch Exchange
PBX (PABX)
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Digital display of an
analogue signals is
used as an interval 0
& 1 and standard in
digital audio and
video
Pulse Code
Modulation
PCM
Telecommunication
public network which
also is called fixed
telephone network
Public Switched
Telephone Network
PSTN
It refers to networks
based on (packetswitched) depends on
mechanism of
controlling powers to
access to appropriate
services
Quality of Services
QoS
Published note about
IEFTF which
explained methods,
approved researches
and innovations
about internet and
systems attached to
it, has unique
number
Request For
Comment
RFC
Standard of audio
and video formats in
internet RFC3550
Real-time Transport
Protocol
RTP
A dedicated protocol
of Cisco is used to
control the network
Skinny Call Control
Protocol
SCCP
ElastixEasy by Haamed Kouhfallah
between work
stations and Call
Manager of Cisco. At
first it was developed
by Cilsios company
and now its owner is
Cisco
13
It is protocol of an
audio signaling based
on VoIP
Session Initiation
Protocol
SIP
It is collection of
signaling phony
protocols which is
used in set up most
PSTN
Signaling System7
SS7
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2. VoIP Hardwares
For installing Elastix telephony system, you may just need a computer
but for communicating with other telecommunication systems or more
comfortable working with Elastix some hardware may be needed. These
hardwares are the most useful ones and can be divided into 3 categories
as follows:
2.1
Access methods and using the services of VoIP
In order to use extension registered on Elastix, some equipment is
needed to register your extension number on it and use it. Generally
communicative ways to VoIP is as follows:
 Analogue telephone adapter(ATA)
This adaptor is known to Gateway. They have network port and placed
on the network with IP. On the other hand they have FXS port which can
be used by connecting analogue phone and extension number registered
on it. These Gateways can have several ports and it is possible to
register an extension number on each port and use it. In the other word
it can be stated that the main task of Gateways is turning an analogue
phone to an IP phone.
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Adaptor of a phone to IPPhone
 IPPhone :
These are similar to ordinary phones but ordinary phone is connected to
RJ11 and in IPPhone it is connected to RJ45 (like network connection)
and all the required softwares and hardwares are Built-in. nowadays this
type of devise is one of the affordable and user-friendly. An SNOM phone
is shown below:
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An IPPhone
 Softphone:
It is software but included in this category. For registering an extension
number you can use a softphone. The biggest advantage is that it does
not cost to you (if you use free versions) and it is easy to use. Eyebeam
softphone is shown below which can be communicated visually.
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Eyebeam softphone
2.2
Computer
In order to install each software you need a computer, Elastix is
collection of the software so we need a computer either. The main
question for bigginers and profesionals is that what properties should we
consider for the sysytem we want to set up? The answer is different
becouse in VoIP, set up and used applications is different. So generally it
can be said that you can chose your system by expiriance and
sometimes you need to put your system in operational enviroment to
see the reproductive rate of system. Some system and reproductive rate
of them is shown bellow:
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Name
City line
Small
City
card(4line),
about 20
extension
number
facilities
features
Voicemail,without 1GB RAM, Dual
recording
Core 2.6 CPU,
conversation
128 GB HDD
Average
E1 line(30
Voicemail,
line), about 50
without recording
extension
conversation
number
1GB RAM,
Core2Duo 2.8
CPU, 128GB
HDD
Average
2line E1 (60
line), about
100 extension
number
2GB RAM,
Voicemail,
Core2Quad 2.8
without recording
CPU, 256 GB
conversation
HDD
Larg
4line E1(120
Line), about
300 extension
number
Voicemail,
without recording
conversation
4GB
RAM,2*Xeon
2.8 CPU, 512
GB HDD
There are some tips about chossing a hardware which is totally
experimental:
 Recording will put extra pressure on CPU. If you want many
extension numbers or to record all of your lines ,you may need the
second server.
 be carefull in using voicemail! This feature will be heavy for your
system either so don‘t active voicemail for the extension numbers
which don‘t need.
 If you want to provide a system, Gigabyte motherboard are better
options especially for installing Linux on them.
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 If you use E1 cards,especially for 2 and more E1 cards, use Echo
Canceller cards with them. They are effective in voice quality and
in reducing extra pressure on CPU.
2.3
Telephony card
Telephone cards usually used for Elastix link with PSTN city lines. Any
telephone card cannot be used for this purpose. It should be Asterisk
compatible.
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3. Installation
3.1
Prerequisites
As it was mentioned for installing Elastix and dialing extention just
a computer is needed. But bear in mind that installing Elastix will
format your computer hard. That computer dedicate to this purpose
and for communicating you need an IPphone which softphones can
be used either. In this article instalation of Eyebeam as a IPphone
is explained.
3.2
Installing ISO
For receiving the latest version of Elastix ISO go to the following
address:
http://sourceforge.net/projects/Elastix/files
for receiving persian Elastix see following address:
http://www.voip-iran.com
installing ISO Elastix is like Trixbox and other distributions of
Linux. For installing, download an ISO file from the site and then
with nero or MagicISO or etc write like a Burn Image on CD and
install it.
Attention: this will format your disk,so be sure you don‘t have any
important information before instalation.
First put Elastix cd on cdrom and boot your system. The first image
which shows installation displayed abit later.
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Installation start with pressing enter, wait for loading files to be
complete, when installer start image of choosing language will appear.
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Choose your language and keyboard language.
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You choosed the language of your system. Then you are welcome to
instalation process. If it was new installation, and there was nothing on
your drivers, you don‘t receive this message. Only when there is
something on the hard and system cannot recognize it,this image will be
shown.
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The next window is driver setting. If you want to install a new system, it
is better to click on remove all partitions, then yes and move to the next
level.
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Attention: if you want to install Elastix just for testing, it is better to
use Sun Virtual Box or something like that. The only restriction is that
you can not use TDM400 card in Virtual Machine otherwise you can have
all the efficincy of VoIP. Bear in mind that network interface should be
bridge.
3.3
Network configuration
After installing Elastix, server get IP from DHCP as a difult. The IP is
shown after entering to the Linux enviroment or with ifconfig eth0, you
can see the IP. Now you can access to the login page of Elastix with
writting IP in addressbar, but if you want to give IP manually to Linux,
there are different ways which the easiest way for begginers is shown as
follow:
system-config-networking
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Chosse the network card for giving IP.
Save the changes and exit. After that you can conneted to UI from any
url with giving IP server. By entering
Username: admin
Password: "Your entered pass on Istallation"
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You can log in. (this username and password is for Elastix version 1.6
and before that. For Elastix version 2 or after that you should enter
admin and your password for installation).
3.4
The default usernames
Elastix version 2 receive passwords of Freepbx, Database, Vtiger,
a2billing, Elastix web during installation while passwords of Elastix
version 1.6 or other programs is as follow:
Web graphical enviroment
Username: admin
Password: palosanto
Freepbx:
User: admin
Password: admin
relationship with customer Sugar CRM:
username: admin
password: password
calculating telephone A2billing:
username: admin
password: mypassword
Vitger:
Username: admin
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Password: admin
Attention: change psswords after installing Elastix as follow:
For changing admin password of Freepbx first respectively go to the
graphical enviroment of Elastix, menue of call center(pbx), pbx
configuration and unembedded freepbx. In this way you will enter to the
freepbx enviroment. Go to the set up, basic, administrators and change
the admin password of freepbx. For other 2 program you need to
change from internal menue of the program.
3.5
Accessing to the graphical enviroment
for seeing graphical enviroment (web) if you set the ip server correctly,
you just need to enter the ip in browser. You can do it with any
computer which is cennected to server through network. It is good to
use firefox for seeing the web.
3.6
changing the admin password
in order to change the admin passwod of graphical enviroment of web,
go to the System and User Management.
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4. Useful programs
4.1
Webmin
Webmin sofware is Swiss knife of Linux!! It means by installing this
program you can easily configure Linux but installation of this program
on Redhat (like operating systems of Elastix and Trixbox) is as follow:
1- First download the file.
http://www.webmin.com/rpm.html
Attention: those who live abroad don‘t have any problem with
downloading from Sourceforge servers and can directly receive it from
Linux.
wget
http://internap.dl.sourceforge.net/sourceforge/webadmin/webmi
n-1.710-1.noarch.rpm
2- Copy the file on /tmp
3- Go to tmp from console:
Cd /tmp
4- Enter the following command:
Rpm –i webmin-1.710.-1.noarch.rpm
The last part of the file is copy so if the version of webmin was different,
the name entered would be different either. After that process the
webmin is installed and for accessing you should enter the following on
your browser:
Http://YOUR-SERVERIPADDRESS:10000
4.2
WinSCP
Maybe you are unfamiliar with Linux or maybe you are familiar but you
are not patient enough to work with that. By the way WinSCP is one of
the best projects of Sourceforge and its task is accessing to the Linux
files. It will be installed on operating system of your windows and Linux
environment can be easily seen, especially it is good for copying the file
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on Linux or changing the files. Professionally it can be said that it is a
FTP & SFTP client for windows.
Those who live abroad can receive the file from Sourceforge.
http://sourceforge.net/projects/winscp/files/winSCP/4.2.8/winscp428set
up.exe/download
Those who live in Iran can see the Utilities of this link.
https://sourceforge.net/projects/vaak/files
You won‘t have any problem in installing it for sure. After that you
should make a new Host for any Linux you want to connect. The page
below is shown after installation:
Now by clicking on NEW the page below will be open. Linux features
should be written in this paged.
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In this page you should fill 3parts:
Host name: IP address of your Linux server
User name: user code for Linux is usually root
Password: password of user
When you make a Host, by clicking on it you will be connected to Linux.
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5. Telephony configuration (PBX)
5.1
General setting:
In this part you can change some of the general and main setting of
Asterisk. Those who are beginners and they just want a simple and small
telephony system, there is no need to change but it is good to know
these features:
 Allow Anonymous Inbound SIP Calls:
The main part of this page is receiving calls without permission of SIP,
which is good to change:
Allow Anonymous Inbound SIP Calls: yes
This item is ―no‖ as the default and SIP inbound calls which are unknown
are not accepted. To enhance the security, it is better to change it to
―no‖ after testing the system.
 Dial Command Option:
In ―Asterisk Dial Command Options‖ you can configure the Asterisk in a
way to show desired performance, for instance if you want the caller
hear the desired song instead of beep sound, change the option –r with
–m (it is not recommended).
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You can use these options if you want:
34
A(X)
Play an announcement for receiver and x
is the played file
C
CDR reset the call
D
It allows the caller to dial one-digit
extension number while he is waiting
for answer. So if that number registered
on context or EXITCONTEXT, it will be
dialed.
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35
D([called][:calling])
Certain DTMF will be sent to the receiver
after answering the call and before the
calls become bridged. ―Called‖ of DTFM is
sent to receiver and ―calling‖ to the
caller. Each 2 parameters can be used
separately.
f
It forced the channel of caller to equal its
caller ID with relative extension of this
channel and for that uses Dialplan hint.
For instance, some PSTN network doesn‘t
allow caller ID to have different numbers
to the caller.
g
When destination channel cut, in order to
Dialplan it continued for the present
extension.
G(context^extent^pri)
If the call is answered, both side of the
call will be transferred by identified
priority. Registration of extension or
context is optional and available
extension will be used if they don‘t
register.
h
This allows the receiver to hang up with
sending ‗*‘DTMF.
H
This allows the caller to hang up with
sending ‗*‘DTMF.
i
If all the required channels are busy, it
will jump by priority of n+101
L((x)[:y][:z])
It limit the call time to x millisecond. It
warns when y millisecond remains and it
repeated any z millisecond. Following
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variables can be used by this item
LIMIT_PLAYAUDIO_CALLER yes/no
(which is used as the default, sound will
be played for callers.
LIMIT_PLAYAUDIO_CALEE yes/no
sound will be played for receiver.
LIMIT_TIMEOUT_FILE it plays after
finishing the time.
LIMIT_CONNECT_FILE it plays at the
beginning of the call.
LIMIT_WARNNING_FILE
it plays as a
warning when y was defined and remain
time will be announced as the default.
M([class])
M((x)[^arg])
Play hold music for caller until one of the
channels get ready to answer.
MusicOnHold can be defined.
Perform the related Macro with the
channel of receiver before connecting
with that channel. Args of Macro can be
sent for putting space between them by
―^‖.Macro can determine the amount of
MACRO-RESULT variable and according
to the amount placed in it, the following
operations performed after finishing
Macro.
ABORT: cut the connection between 2
callers
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CONGESTION: it seems the line has
congestion.
BUSY: when the line is busy. It causes
that the program jump by the priority of
n+101(if the j was determined).
CONTINUE: cut a call of receiver and
allows the caller to continue Dialplan
performance and go to the next priority.
GOTO:<context>^<extent>^<priority>:
transfer the call with identified priority
and can clear the extension and context.
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n
This option changes the Screen/Privacy
and clear that there shouldn‘t be any
introduction in directory of privcallerintros.
N
This option changes the Screen/Privacy
and identified that caller ID is available
and there is no need to screen the call.
O
It changes the caller ID of channel of
caller to the caller ID of channel of
receiver. Asterisk version 1.0 and before
that were capable of doing this.
p
It activates the screen mode option
which doesn‘t have protection like
Privacy mode
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5.2
P([x])
It activates the privacy mode and if x
was defined, it was used for
Family/Privacy key. If it wasn‘t defined,
present extension will be used.
r
It plays the beep sound for the caller and
there is no sound before answering.
S(x)
Cut a call x second after answering the
receiver.
t
It allows the destination to transfer the
call by the process of DTMF defined in
features.conf
T
It allows the caller to transfer the call by
process of DTMF defined in feature.conf
w
It allows the destination to record the
conversation by DTMF process which is
defined for one-touch recording in
features.conf file.
W
It allows the caller to record the
conversation by DTMF process which is
defined for one-touch recording in
features.conf file.
Extensions
You can dedicate each number you wanted to your extension. Just don‘t
use the following:
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79-70
They have been booked for
holding the calls
799-700
They have been booked for
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holding the calls
7777
It has been booked for
simulation of the incoming calls
911
Emergency call (in Iran is 110)
999
Emergency calls of other
countries
5.2.1 Create extension
For making extension the following menu is used:
You can choose a Protocol and create the extensions:
Generic SIP Device: SIP is a standard protocol for ATA & VoIP phones.
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Generic LAX2 Device: LAX is used for connection between two servers
of Asterisk or you may want to have IAX extension.
Generic ZAP Device: ZAP is a hardware which is connected to your
Asterisk server and they are used for defining the FXS modules.
Other (Custom) Device: this term is for non-standard devices like
H.323. It also can be used for mapping of an extension to another
number.
For creating an SIP extension, Genetic SIP Device should be chosen and
then we click on submit. Enter your information and click on submit:
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User extension: dedicated extension of the user
Display name: (optional) name of user
Secret: password with extension number used for authentication of the
user.
Don‘t forget to click on Apply Configuration Changes.
5.2.2 ZAP extension
Creating ZAP extension (for module of FXS) doesn‘t have any difference
with creating SIP or LAX extensions. You should just use ZAP Genetic
Device instead of SIP Genetic Devise (bear in mind that newer versions
of Asterisk and Elastix for recognizing hardware use DAHDI instead of
ZAP but the word ZAP is used for recognizing the hardware).
Attention: it is important that enter the channel number of your module
properly otherwise the sound will be sent one sided or fax devices
attached to the FXS module will have trouble in sending fax.
Attention: consider that if you send sound on SIP, you should use ulaw
and alaw codec because the fax which is sent by other codec will be
failed.
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5.2.3 Enable voicemail
During registering extension you can enable your voicemail otherwise
whenever you need to have an extension and voicemail you can enable
it. For that you can go to the phone setting of extension and enable
voicemail on any extension and click on it (create a new extension).
Voicemail password: the password of your voicemail
Email Address: email of a person who has that extension (it is
recommended if you want to be informed through emails)
Email Attachment: yes (attaché your voice mail in email)
Play CID: yes
Enable Envelop: yes
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VM Context: default
After changing these, click on submit and Apply Configuration Changes.
5.2.4 Definition of Extensions Batch
In this section you can enter extensions batch to the system by excel &
…. In order to work easily with this part, it is recommended to create
one or two extension as it was said.
Then click on file of extensions batch and choose to download extensions
in CSV format. You can download the extensions file on your local hard.
After opening the file with your favorite program, you can see the name
of columns and extensions you have created. By that you have a guide
to fill your file cells. Most fields are repeated in all users and you can
copy them and fill the sections you need.
After entering all the extensions, save the CVS file with your desired
name. Now you need to upload your file in server. You should enter to
the extensions batch files and click on brows. Choose the file, where it is
saved, and click on CVS upload file and that‘s it! For checking extensions
go to the phone setting and see your extensions.
5.3
setup softphones:
 Setting the CounterPath Xlite:
First download this softphone from CounterPath or VoIP-IRAN:
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http://www.counterpath.com/xlitedownload.html
eybeam software is one of the best softphones of the world which is
used for SIP calls. This is made by Counterpath Company and its free
version is X-Lite.
This software gives the following option to the users:
 The application provides a graphical interface which is made by
phone standards.
 Feasibility of using 10 account of SIP simultaneously.
 Having video calls.
 Providing call transfer
 Recording the conversations
And so on.
After installing the program and entering the serial number, you should
create an SIP account. (Bear in mind that this account should be made
in Asterisk, Elastix or any other server which you used). For that you
should click on show menu above the eyebeam.
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After clicking this window will be open:
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Choose the SIP Account Setting. The window will open as follow:
Click on add key and the windows of properties of account will be open:
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As you can see there is different part in this window that should be filled
with information given to you by network manager.
Display name: when you call someone, receiver sees your name and
number. Enter the name you want to see.
Username: enter your username. Generally the user name is the same
with your extension number.
Password: enter your password.
Authorization user name: again enter your username (this part is
used for connecting to Asterisk by Sip Proxy and if you are connected
directly, it is not necessary to enter that).
Domain: enter your name or ip server. (Just enter your name when you
have DNS Server in network)
For other parts don‘t suffice to default setting and click on ok. The
following window will be open and your account will be shown there.
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Click on close. Eyebeam tries to connect with server (be careful in some
systems firewall ask you if you allow eyebeam to access or not. You
should let eyebeam otherwise you will have problem in receiving or
creating calls). After connecting eyebeam with server and registering
your account in server, eyebeam shows the ready message.
Some of the common errors of eyebeam:
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Error 403: trouble in registering: it occurs when user make a mistake in
creating SIP account and by checking username and password and… and
reentering them it will be solved.
Error 408: request timeout: it occurs when eyebeam face one of these
problems:
 Problem in network that cause not to find the server.
 Mistaken ip or name of the server which entered in SIP account
section.
 Closing the port of SIP by firewall on the server or system of the
user.
Not hearing the other voices: it can appear when:
 Defected headset or not being connected to device (it is obvious
that if the headset wasn‘t connected, no voice can be heard. You
should choose the easiest way to solve the problem if you have the
network management)
 If the speaker‘s voice was closed in Eyebeam.
 Using the Codec that is not activated in Eyebeam. For managing
the Codec in Eyebeam you should go to the Show menu--> Options
--> advanced--> Audio codec and chose the activated codec in
server.
Not being able to hear your voice:
 Defected headset or not being connected to device (it is obvious
that if the headset wasn‘t connected, no voice can be heard. You
should choose the easiest way to solve the problem if you have the
network management)
 If the speaker‘s voice was closed in Eyebeam.
 Defected Microphone or not being connected to the device.
 If the key of microphone was inactive on headset. (The most
common problem happened in our company). When the
microphone being muted on Eyebeam.
 Lack of coordination of Codecs
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Eyebeam has a secret part for hiding the advanced setting which is
possibly good for professional users. Dial ***7469 for entering to this
part.
5.4
follow me
When we registered extensions, Elastix can be set to meet our needs. It
is possible that we want the system automatically connected to the
extension we already defined if our extension didn‘t reply. And we
should do as following:
Call center, configuration of telephony system, follow me
We faced with this window:
Choose the extension you want to define this features.
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Ring Strategy: ringallva2 dial the main number first and then the
others.
Extension list: 1102 is the deputy director and 1103 is the office
assistant.
Ring Time: 20 second
Destination If no answer: terminated call-hang up
Whenever dialed to the manager, dials the Asterisk extension number of
the manager. If no one replied, the contact is with the extension of 11
and 22. And if no one answered again, Asterisk terminates the call. After
finishing, choose the submit changes key and Apply Configuration
changes here.
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Attention: as we saw in follow me, several extensions can be determined
as the next destination and it is the main difference between follow me
and call forward.
5.5
What is dial pattern?
Dial pattern is a pattern for management and supervision of Asterisk for
sending and receiving the dialed number or entered on telephony
system. For instance a number is dialed on the system by a user; you
can manage the number with a dial pattern, add or remove something to
it. It is used both in Outbound & inbound route and in Trunk. Basically
you will face with dial pattern in any part which deals with call
management. There are some rules for definition of a pattern as follow:
X: represent a number between 0 to 9
Z: represent a number between 1 to 9
N: represent a number between 2 to 9
. : When there is a point in the pattern it means that there can be more
number after that.
|: it means the number before that should be removed.
[ ]: if you want to choose your number, for example [1356] it means
one number between 1, 3, 5, and 6. Or [3-7] means a number between
3 to 7
For example:
9|X.| it is a popular pattern. It means choose the numbers start with 9,
take the 9 and send the rest to the destination.
ZX this pattern means the dialed numbers which are 2-digit, and their
first number is between 1 to 9 and their second number between 0 to 9.
NX.
It means 3-digit number and more that their first number is
between 2 to 9 and rest can be anything.
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5.6
What is trunk?
Trunk is used for connecting Elastix with outside and other systems. For
making trunk required protocols for connecting with outside and other
system should be determined and define trunk according to it. The
current protocols are:
Zap: it is used for installing the Asterisk cards (Dahdi support) on
telephony system. After putting the card in server, a Zap trunk should
be defined for sending and receiving the calls. As a default a Zap trunk is
defined in Freepbx known to ZAP/g0 which there is no need to install a
card on system. But if you want to install more than one card, you
should another ZAP trunk.
LAX: it is used for connecting to Gateways and VoIP providers with LAX
protocol. One of the important uses of trunk is connecting 2 Asterisk
servers together which are discussed later.
SIP: it is used for connecting to Gateways and VoIP providers with SIP
protocol.
ENDUM: its full statement is E.164 Number to URL Mapping. And
translate the numbers to the internet addresses.
DUNDI: its full statement is Distributed Universal Number Discovery. It
is a mapping system on VoIP network which any group knows the
channel of phone group around on the network (at least one number)
and it is created by Mark Spenser creator of Asterisk.
Custom: for the other Trunks that Asterisk support or protocols such as
h323 or SCCP that added you may define several trunks. For example
you define a VoIP trunk for international calls for a service provider and
a PSTN ZAP trunk for connecting to city line through card. For defining
any trunk, especially VoIP trunks for connecting to a Getaways or VoIP
provider, you should request the configuration and definition of the trunk
from the provider or manufacturer. They could be different. Here we
have a case for example. The case below is for connecting to the
Pennytel provider.
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Click on SIP creation of the trunk; consider an outbound caller ID, of
course you should follow the general format; it should have the name
and number as follow:
―Your name‖<your number>
Maximum channels: use it for the emptiest capacity unless you want to
limit. For instance for creating an urban ZAP trunk, you have a 4 port
urban card but it create a trunk with 2 channel that only two call can be
mapped to the trunk simultaneously.
Dial Pattern: it was discussed earlier that before manage the numbers
before sending them to the trunk. For example adding or subtracting 9
to them. But it is popular in outbound trunk.
Peer Details: is some setting for sending call. The following setting is
shown for sending a call to Pennytel provider.
Username: 8880XXXX
Type: peer
Secret: password
Insecure: very
Host: sip.pennytle.com
Dtmf mode: rfc2833
Disallow: all
Allow: allow&ulaw&gsm
Canredirect: no
Canreinvite: no
User Context: it is the user number for
usually don‟t use for connecting to Gateways.
connecting.
User Details: required setting for receiving the calls
Canreinvite:no
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It
Context:from-trunk
Fromuser: 888XXXXX
Qualify: no
Secret: password
Type: user
Username: 888XXXXX
Registration: some service providers want this field:
888XXXXX: password@sip.pennytle.com/888XXXXX
For more information, see the links below:
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
http://www.freepbx.org/support/documentation/moduledocumentation/trunks
5.7
Outbound Routers:
Basically routes are used for mapping in Elastix.
The purpose of
Outbound routers is that a destination determined for each number
gotten through ATA,IPPhone and Softphone. As a default of Asterisk if
the number dialed, it goes to the extension. If there is no number with
that extension, outbound route find the destination.
The most widely used of outbound routes is removing the number 9 and
sending the rest to the urban trunk which is defined as a default due to
the high usage. For defining the outbound route go to the call center
menu, telephony system setting and click on outbound route.
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There are 2important setting and definition of route:
 Dial pattern which was mentioned fully in previous chapters. In this
example you can see that this route is for calls that start with 9 and
Asterisk took the 9 according to the dial pattern and sent to the
determined trunk.
 Trunk sequence determine the trunk for this route and you can
relate several trunk to it, so trunk are in the rank by priority and if
the higher trunk is wrong and Asterisk cannot guide the calls, it
moves to the next trunk.
For example you defined an outbound route for 00 and
international calls, in this case you may use different service
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providers of international phone. Trunks put them in priority in case
to failure to communicate, use the alternative trunk.
5.8
inbound routs:
The outbound calls to the Elastix should have an inbound route in the
system otherwise Elastix wouldn‘t accept that. So choose an inbound
route from call center and telephony system setting.
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In the beginning of configuration, you can determine that this incoming
router goes to lines (DID Number) or telephone number (Caller ID
Number). For example if you use E1 for urban lines, you can determine
an inbound router for main DID or define an incoming router for especial
numbers (Caller ID) like the calls with pre number4465x connected to
the announcement (especial message play for them).
Attention: you can use the dial pattern for the DID and Caller ID. When
you write 4465X in all Caller ID, it include all caller IDs with pre number
of 4465.
At last a destination should be determined for identified calls. You should
set your destination in that field. Bear in mind that defined items are
shown here. For example if you don‘t define any announcement, there is
no option for that.
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Attention: if you don‘t need to separate the incoming calls or you just
have one urban trunk and want to direct them to the specified
destination, you don‘t need to define DID or Caller ID and keep the fields
empty. Just determine one destination and click on submit. In this case
system creates an outbound route, Any DID/ Any CID, which include all
the incoming calls to the Elastix.
5.9
Time Conditions
This feature is used for this purpose that you could use several pattern
in different time for inbound routes of system. For example you want
IVR numbers 1 in working hours and IVR number 2 in closure of the
company respond with 2diffrent algorithm.
To use this system in two parts the following settings must be applied:
Definition of response time:
In order to define in ―call center‖, telephony system configuration of
menu, choose the option of response time conditioning and click on Add
Time Condition.
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First, specify a name for it, in second field you should choose the
determined time group. Time group is a specified interval such as
month, weeks, days and hours and by choosing a group you determined
that this situation belong to which interval.
Attention: for specifying the interval of time group, there is no menu in
Elastix. It should be done through Freepbx. So from the menu go to call
center, telephony system setting and choose Unembeded PBX. For
entering to freepbx, use your admin user and password. (if you use
Elastix version 1.6, the password is admin)
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In freepbx, choose the time group.
For example in above figure, we determine the group for working hours
-8 am to 5pm- and the whole weeks of the month. Just consider that if
you want to have plan for other hours, you should determine a time
group for them either.
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Now back to the time condition, you see a defined group in list of time
group. Also In the lower part you specified the destination if time
matches and if it doesn‘t.
5.10
system recording
In some part of Elastix you can add recording to the system and use
them in menus and different system format. This part is in call center,
telephony system setting in menu of system recording.
In this way you can transfer audio file to the telephony system. The first
one is recording extensions and it lower the sound quality but it is
essential. So enter your extension number to the specified place and
after clicking on Go, your extension will ring. After recording you can
name and save it. But if you want to put a prepared audio file on your
system, choose your file on browse and upload it. After that you can
name it on Elastix.
Attention: your audio file should have this format for uploading:
PCM Encoded, 16Bits, at Mono 8000Hz.
5.11
final destinations
Any call from trunk that enters to the Asterisk should have a destination
by Asterisk which is called final destinations. These destinations can be a
simple extension, a complicated IVR or a written program by you. There
are some common final destinations:
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5.12
digital receptionists (IVR)
(IVR) interactive voice response is said to digital receptionist. An IVR
plays the recorded text to the caller and ask them to press the key to
connect to an organization, work group, a person or etc. then IVR send
the call to the destination. It accepts any key to determine the call
destination (ring group for the sale). In addition, 2 options of ―I‖ and ―t‖
can be defined. ―T‖ is used for timeout and ―I‖ for incorrect input. For
having digital receptionist, you can click on IVR link on the left side of
the page.
2part of setting of digital receptionist is shown. IVR options which is
shown in the figure below with 12 adjusted options.
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Change name: this option is for recognizing the menu and is not
translated by freepbx.
Announcement: it is used for recorded messages for IVR. It should be
recorded already and added to the message list of freepbx from system
recordings.
Timeout: the entered number here is equal with the time that IVR
waited for pressing the key by caller. If it doesn‘t receive any, the call
will be transferred to the ―t‖ part and the call will be terminated.
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Enable directory: If this option is selected, it allows the caller to press
―#‖ key and entered to the directory system and search by name. By
pressing the key pound, it plays a message to the caller that contains
the entry stages.
VM Return to IVR: if it chooses, the transferred call from the IVR to
the voicemail turns back to the IVR finally, after finishing with the
voicemail and messages. If you don‘t select this option, the call will be
terminated.
Enable Direct Dial: by this the caller can connect directly with dialing
extension. Otherwise by entering the extension-even it exists- you hear
that your number is wrong.
Loop Before t-dest: if you select this option, after considered period
for timeout, menu will start playing again. It will be repeated with the
specified number in Repeated Loops. If you don‘t select this option, the
caller will be transferred to the determined destination in ―t‖
immediately or the call terminated.
Repeat Loops: the number of iterations of IVR
Timeout message: the message played for the caller when no key
were pressed after the time entered to timeout. And it will be played
when there is no destination for ―t‖.
Loop Before i-dest: its performance is like loop before t-dest and
check the key entry error. If you don‘t choose this option, the caller will
be sent to the determined destination in ―I‖ or the call terminated.
Invalid Message: it is the message played when the caller make a
mistake in IVR menu. Consider that it will be played when there is no
destination for ―I‖.
Repeat Loops: it determined the number of repetition of voice menu if
there was an incorrect incoming and after that the call will be
terminated. The maximum repetitions number is 9.
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The next part is IVR destinations that mapped the pressed key of caller
to the intended destination. It is as follow:
In this example, key 1 sends the caller to the queue of technical
support. ―t‖ destination terminates the call an ―I‖ destination is
transferred to the voicemail of 501. The options can be increased by key
of Increase Option. An IVR can have multi-digit options. Consider that
IVR wait a while for one-digit incomings (for example, when a caller is
allowed to enter the 1, 10, 11, 12 options, the user should wait for
option 1 to be sent to its destination).
5.13
Ring Group
Ring group is a group of extensions that ring by external incoming calls
simultaneously. You can add your mobile number if you want (you
should have a trunk and a route, if you want your mobile phone
ring).you can have a ring group for per incoming trunk or have one ring
group for all trunks. For start we assume that we just have one ring
group which is for all incoming trunks.
We consider the ring group 33 which is including project manager,
network manager and deputy manager (one-digit numbers are not
recommended for ring groups; try to dedicate at least two-digit
numbers). If none of the managers reply, Asterisk will terminate the call.
Use this way to enter to the ring group:
Call center, telephony system setting, and ring group
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After entering to this part, you will see this page:
Click on add ring group.
The options are:
Ring-Group Number: it is the number for accessing to the ring group.
It is like extension number, both endpoints and users can dial it.
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Group Description: it is used for recognizing the ring group and it
doesn‘t have more efficiency than that. The users cannot be seen by
callers.
Ring Strategy: it determines that how Endpoints should call. Endpoints
will be ringing by one of the following ways:
Ringall: all the groups determined in extension list can ring at one time
and together.
Ringall-prim: in this case if the first number in the list wasn‘t busy, all
the numbers would start calling. If the first extension was busy or be in
the do-not-disturb mode, none of the extensions ring and the call
directly sent to the determined part in destination if no answer.
Hunt: if one of the group member replies, the others that specified in
Extension list field starts ringing until one of the members answer.
Hunt-prim: it‘s the same with ringall-prim. If the first extension was
busy or was in the do-not-disturb mode, none of the extensions would
ring; otherwise it has the same process as hunt.
Memoryhunt: the first extension in the list rings and if there was no
response, the second extension would start and similarly to the end
(until one of the extensions reply or be timeout).
Memoryhunt-prim: if the first extension was busy or be in the do-notdisturb mode, none of the extensions ring otherwise it has the same
process memoryhunt.
Firstavailable: the first available extension in the list will ring. If call
waiting activated for the extensions, it would be consider available,
however it is on another call.
Firstnotonphone: the first available extension in the list will ring. Call
waiting setting is not consider here. If one extension was busy, without
considering the activation of call waiting, press the submit changes key
and Apply Configuration Changes Here
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5.14
Queue
What is queue
5.14.1
The meaning of queue is the same that we all know. Whenever your
request is more than the resources or service providers, you need a
queue. In the other word, in call center if you want to reply to the callers
with limited number of receptionists, you need a queue. When the
receptionists are busy, you keep the callers in queue until one can
answer. This is the main task of queue.
5.14.2
Kind of queue
There are 2kinds of queue: static queue and dynamic queue. This
division is not because of their function but it is according to those who
are responsible to reply. In static queue, the extension in queue is
responsible to answer and connect the incoming calls queue to the
defined extension. In dynamic queue, the agent is responsible to answer
instead of extension. You can have a static queue call center with
Elastix, definition of queue and extension but for having a dynamic
queue call center you need a call center module of Elastix in addition to
definition of queue because it makes up the necessary equipment for
defining agent and working situation. By the way for having a call
center with any queue, you need its definition and familiarity with its
capabilities. In next chapter, we will discuss more about the queue and
its performance.
5.14.3
Making queue
Whether you want to have PBX or any kind of call center; you need to
define a queue. It is one of the powerful features of the Asterisk which
can consider a powerful tool for selling products. Facilities of queue are
so complete that can set up an expensive call center and Enterprise and
sell it.
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In setting page of queue there are lots of features and parameters that
some of them may be useless but in this chapter we try to explain then.
At first we deliberate on Add Queue.
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The name and the extension to reply are determined.
Queue Name: a name for queue but try to choose a name
associated with the purpose of the queue such as selling queue
Queue Password: this is for agents who deal with dynamic queue in
call center, so if you want if you want to work without call center
module of Elastix and have a call center with this queue, don‘t fill it.
CID Name Prefix: you can add a prefix to the number of caller for
example queue name. In this case if your agent was the member of
some queue, can be found from the number that this call is
transferred from which queue.
Static Agent: you should write the extension numbers that are
responsible for answering this queue. You can also help from
Extension Quick Pick for choosing extensions.
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You should consider that if you want to use call center module, the
number of defined agents should be written in that module which will be
explained later.
Attention: when you use Extension Quick Pick for your extensions, it
added a ―0‖ number in front of the extension, writing that number is not
required. That number is fine that its use is not mandatory. You can put
a fine number. The more the number was bigger, the more calls will be
connected to that extension.
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 Queue Options
The most important and most attractive part of a queue! Each of
these parameters can take you close to a big contract so each
function will be always remembered.
- Agent Announcement: playing a message for extension or agent
before connecting call. It is possible that your agent be a member
of some queue; by leaving a message for each queue such as ―this
call is from queue sale‖, then your agent know the incoming queue
before getting the call.
- Join Announcement: playing a message to the caller upon
entering the queue. It will be played once. It can be a welcoming
message like‖ hi, welcome to the sell part of X company, please
wait for connecting a call‖.
- Music On Hold Class: determining a music for the Callers who are
waiting in queue. This is the chosen music for Music On Hold of a
system as a default.
- Ringing Instead of MoH: the ring sound will be played instead of
music for those who are waited in queue.
- Max Waited Time: the maximum time that a caller waited in
queue. When it finished, the caller
Will be out of queue and sent to the determined destination in Fail
Over Destination.
- Max Callers: determining the capacity of queue for admission. For
example you consider the capacity of your queue 10, the 11th caller
is not allowed to enter and after hearing the message of full queue,
come back to the queue of previous menu.
- Join Empty: default of this option is ―yes‖ it means if the defined
extensions and agents were not activated, should the caller enter to
the queue? But if you don‘t want the caller enter to the queue when
there is no activated receptionist in queue, change this option to
―no‖.
- Leave When Empty: it is the same with join empty but there is a
difference that in this case if someone waited in queue and suddenly
the agent was not available, the waited one should be thrown out of
the queue or not.
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Ring Strategy: This option includes algorithms for how to connect
the caller to the extensions of the queue.
Ringall: all the groups determined in extension list can ring at one time
and together.
Ringall-prim: in this case if the first number in the list wasn‘t busy, all
the numbers would start calling. If the first extension was busy or be in
the do-not-disturb mode, none of the extensions ring and the call
directly sent to the determined part in destination if no answer.
Hunt: if one of the group member replies, the others that specified in
Extension list field starts ringing until one of the members answer.
Hunt-prim: it‘s the same with ringall-prim. If the first extension was
busy or was in the do-not-disturb mode, none of the extensions would
ring; otherwise it has the same process as hunt.
Memoryhunt: the first extension in the list rings and if there was no
response, the second extension would start and similarly to the end
(until one of the extensions reply or be timeout).
Memoryhunt-prim: if the first extension was busy or be in the do-notdisturb mode, none of the extensions ring otherwise it has the same
process memoryhunt.
Firstavailable: the first available extension in the list will ring. If call
waiting activated for the extensions, it would be consider available,
however it is on another call.
Firstnotonphone: the first available extension in the list will ring. Call
waiting setting is not consider here. If one extension was busy, without
considering the activation of call waiting:
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Agent Timeout: the frequency (second) that the extension of the
agent rings until the system timeout.
Call Recording: recording the conversations of the queue. Audio files
will be saved in /var/spool/Asterisk/monitor. Never forget that
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recording the put extra pressure on server, so select a server
capacity in terms of both processing power and disk space.
Queue Weight: you consider a weight for a queue. You extension
could be in extension of some queues simultaneously. So the weight
of queues determines the priority for sending a call to an extension.
 Caller Position Announcements
- Frequency: determining the period for playing a message in
second (choosing 0 for inactivation)
- Announce Position: declaring the position of a person in queue for
example ―you are the 5th person in the queue‖
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Announce Hold Time: declaring the estimated time for waiting. If
it was less than 1minute, it wouldn‘t announce. If you choose once,
the message will be played just once.
 Periodic Announcements: you can choose an established IVR in
menu of IVR to be played for people in queue alternately. It can be
used in this way: for example you want that an IVR played for the
waited people who say ―for returning to the main menu press key
1, otherwise wait‖. You can play any IVR for people in queue.
Fail Over Destination: if for any reason the queue functions
wrong, you can send the people in queue or those wanted to be, to
the final destination and don‘t miss any call. Bear in mind that in
final destination, you always will see the items in a list you‘ve made,
for example if there is no announcement on your system yet, you
won‘t have it on destination.
5.15
Announcement
Sometimes it is necessary for you to play a message for the caller in
telephony system, for example you want that if someone presses key
8 on IVR, the address plays for them or agents play an address or a
message with transferring a call to a number so they use
announcement.
In menu in ―call
―announcement‖.
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system
setting‖
choose
At first choose a name for it. The other configuration is:
Recording: from list choose a message which was loaded through
menu of recoding.
Repeat: choose a key to repeat a message when a caller entered.
Allow Skip: if it was activated by pressing any key the message can
be cut and directly send to a next destination.
Return to IVR: if the caller was sent by an IVR to this message, the
next determined destination (last option) cancelled and the caller will
be back to the IVR. So we determine in Destination after playback
that after playing message caller directed to which destination.
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5.16
Call back
A callback will hang up on the caller and then call them back,directing
them to the selected destination. It is used in situation that the caller
cannot access to the VoIP endpoint and don‘t want to pay the cost of
long distances. This is useful for reducing mobile phone charges while
inbound calls are significantly cheaper than outbound calls.
Destination of callback can be any resources defined in PBX (like
extension, voicemail, IVR, queue or…) or be used like a complex with
DISA, which explained later, that caller receive a beep sound and be
able to dial.
For having a callback click on callback in internal option and
configuration and you should fill four options in next part.
A)
B)
C)
D)
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Callback description
Callback number
Delay before call back
Destination after callback
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In callback description a name is determined for call back setting.
In call back number a telephone number is put that a system dials it for
connecting a call. But you should consider that the number format
should be the same with one of the formats of call back (for example if
you don‘t have any format or pattern for 11-digit numbers and enter
09121111111, callback setting doesn‘t work and assumed as incomplete
number). If you put this part empty, the number of the caller ID of the
caller is used for calling (be sure that the caller ID be correct and don‘t
registered like ―unavailable‖ or ―unknown‖.
If you set a time in Delay Before Callback, there will be a delay (in
second) between cutting a call and the process of starting the next
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callback. It is obvious that this part is optional and you can leave it
empty. But consider that most of the telephony systems have some
seconds wasting time to be ready for call back after cutting a call, so it is
good to wait some second that the phone (mobile) accept the next call
and not to be busy. Destination after callback is used for setting a call
target that should transfer the caller to it after terminating the call. In
this part any call destination is usable.
After setting click on submit changes and apply configuration changes.
So the call back is saved and activated.
A few cases of application of the callback can be summarized in the
following:
 Consider a company that its employee could access to the
voicemail from everywhere. Calling with toll-free of company for
checking the voicemail cost a lot for company. In this case after
hanging up on any caller, call them back and direct the
conversation to the (Misc Destination) 98*. The callers receive a
call from the number they dialed and they extension number and
the password is asked from them, so they can access to their
voicemail.
 There are some companies that the cost of their VoIP trunks of the
calls is cheaper than calling with mobile of staff with company.
Mobile of the staff doesn‘t have the fee of dialed calls. In this case
a callback is set for each of employee that after cutting the call, call
back to their mobile phone and DISA destination give a beep sound
from PBX; in the other words, by dedicating one of the VoIP trunks
of a company, he is allowed to call with no cost with his mobile
phone. In next chapter we will discuss fully about DISA.
 Large companies need to collect calls from all over the world (for
example a credit card company that needs to get all the missing
contact card, forgetting the password or stolen password). By using
the VoIP trunks of the country that caller called it reduce the
international cards significantly. In this case it calls back to the
caller and put him in the call queue. Callers receive a call from the
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number they called and connected to the representative of the
company once prepared.
5.17
Accessing to dial tone from outside (DISA)
DISA is used when a free line (beep sound) is dedicated to a caller from
PBX. By having beep sound a caller can use all the accessible endpoints
of VoIP. It means that a caller that hear beep sound can dial any
extension he wants, check the voicemail or even dial from the attached
line of PBX. For having DISA or the service of accessing to dial tone from
outside, click on menu in left side of the page on the DISA, below the
internal option and configuration as follow:
In the following eight options will:
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DISA name: it is used for specifying the settings of DISA in different
part of system as call destination.
PIN: it is used for recognizing the caller when accessed to the DISA call
destination. If it is not empty, it will ask to enter pin code from the
caller. If it isn‘t equal with the pin, the call will be terminated and the
caller is not allowed to access to the DISA call destination. By the way,
PINs can be separated by comma. (For example by entering 9012, 5678,
1234, caller will hear beep sound by using each of 9012 & 5678 & 1234
numbers).
Response Timeout: it determines the time (in second) that system is
waiting for receiving an entrance from the caller before terminating the
call. This time is for entered and to be entered numbers. This time is
considered 10 second as a default but it is not enough and the caller will
be in hurry. 15 or 20 second is enough for entering the numbers without
waiting too much.
Digit Timeout: it determines the time that system should wait between
entering the numbers and dialing. It means after entering the numbers
by caller and the time he is waiting for connecting, how long (in second)
a system should wait to send the numbers to Asterisk for connecting a
call? The default time is 5second and it is enough for the callers to enter
the numbers. The caller won‘t last more than 5second for entering the
numbers.
Require Confirmation: when this option selected, system ask the
caller to press key 1 whenever he hear beep sound. If it doesn‘t, the call
will be terminated. This option is usually used when the caller is sent to
the DISA from inbound route directly. In this case the caller immediately
will be sent to the destination without any beep which timeout quickly
and the call terminated without the caller being aware of what
happened.
Caller ID: it is used when we want to consider that the outbound route
be something other than caller ID of the caller for the DISA call
destination. The format of this option should be like <##########>
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―caller name‖ that instead of the caller name should be the name of the
caller and instead of <##########> telephone number of the caller
(for example <09111234567> ― Saeed‖). If the option was empty, the
number of the caller should be sent for destination as Caller ID. (Likely
the number that you put in this option won‘t be sent to destination. So it
is better to leave it empty, this setting depends on your trunk, for more
information refer to ―trunk‖ subject).
Context: it determines the context that Asterisk put the caller in it in
DISA calling. (More information)
Allow Hang up: if it was chosen, the caller can enter the terminating
code (** as a default) and cut the call and in this case hear the beep
sound. It can assure the caller that the call is terminated.
After finishing the setting, click on submit changes and for confirming
(click on orange border of DISA) apply configuration changes and it is
activated.
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6. Fax Server
As you know Elastix have a good and powerful server for sending and
receiving fax. This fax server use Hylafax software. Elastix can receive
faxes from difference ports such as LAX, SIP & ZAP trunks.
6.1
Making a IAX Extension
First of all you should define an IAX extension. This extension is like a
modem for connecting Asterisk and Hylafax. For any fax server you need
a lax extension so go to the call center, telephony system setting, and
extension.
Choose Generic LAX2 Device and click on submit.
The only setting that should be entered or changed is as follow:
User Extension: extension number
Display Name: a name
Secret: extension password
Record Incoming: put it Never
Record Outgoing: put it Never
Now click on submit, your extension or your modem is ready!
6.2
Visual Fax
In second step you should make a visual fax. Go to this menu. Fax,
visual fax. There is some information required for making a visual fax
which is your fax server either:
Visual Fax Name: “a name‖
Visual Fax Name (LAX): ―The lax extension you made‖
(LAX) password: “the lax password you made!!‖
Destination Email: “the email you want to use for sending fax to it‖
Code of Country: “can be 98‖
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Code of Area: “can be 021‖
Caller Name: “name of your company‖
Caller Number: “phone number of your company‖
After making this, go to the list of visual fax. That should be shown here
and Running and idle on ttyLAX1 should be written in ―position‖ field.
Now your system is ready to receive fax. For testing you can call to the
lax extension and hear the sound of fax. On your IVR you can consider a
key like 6 for the fax, anyone press the key 6 will be connected to your
fax. You can use this way for making more faxes and have several visual
fax system.
6.3
View the Faxes
Received faxes are like PDF and after receiving in fax menu, view the
fax can be seen.
6.4
Programs of sending fax
 Sending fax by Winprint HylaFAX:
Winprint HylaFAX is a windows GUI interface for sending fax
through server. By using this software you can fax your data, as
easy as printing, with giving destination fax number. For receiving
this software please refer to VoIP-IRAN in sourceForge, files,
utilities and fax.
https://sourceforge.net/projects/vaak/files
By installing this file port of WinPrint HylaFAX is added. After setting
up the exe, go to control panel, add printer and click on printer add a. I
the page below choose Local Printer but since it‘s not a printer don‘t
choose install my plug and play printer automatically detect and.
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Click on Next and in Select printer port, choose create a new port
type. If it is successfully installed, you should see the Winprint Hylafax in
the list. Select it and click on Next.
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The naming box will appear as above. The name is not important but
it is better to use :HFAX1 to determine it.
Choosing the type of printer, as it was shown choose Apple
LaserWriter 12/640 PS from Apple.
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You can name whatever you want.
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Don‘t share it. You can install Winprint HylaFAX on any system and
directly connected to the server of Hylafax. Don‘t print the test page
because you have not set the printer yet. When you install it, right click
on it and choose properties. Find the tab of ports and tick the port you
have made.
Click on configure port for connecting with fax server.
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Enter IP of the server in hylafax server address and write the
username and password of the server. One of the problems that may
occur is not having permissions to access the hylafax server. To solve
this problem go to var/spool/fax/etc/hosts.hfaxd/ and add IP of your
computer. Now you are ready to send a fax. For testing choose any
application you want and click on print and choose your printer,
winprinthylafax. Whenever you send something to this printer, the page
below appears:
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Enter the fax number totally. Fax will be sent by halafax and its server
send an email to confirm the address entered to Notify E-mail.
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7. Callcenter module
7.1
introduction
Call center module is one of the add-ons of Elastix that can be
installed optionally. This module allows having a call center with dynamic
queue by defining an agent. This module has different part and services
such as browser page of agents, do automated calls advertising
(Telephone marketing), Very detailed reports of agents, their operation
and performance and many other useful items that will be explained in
next chapters.
7.2
Concept of call center
Generally the concept of call center is systems that can service
callers with limited number of receptionist. Normal Pbx systems are
in this category. The most important feature that distinguishes a
center with a normal pbxs, is Queue. It allows the system to service
callers with much greater number of receptionists.
7.3
the
not
call
the
installing call center module
After installing Elastix go to add-ons menu and wait until it connects
to the Elastix. Note that it may take several minutes; by the way be sure
that your server is connected to internet. After connection click on
Install.
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After finishing the installation you can see the added menu of call center
in main menu.
7.4
use of call center
Call center module has two applications:
 Management of ingoing calls: it is receiving call from queue and
accountability of it to agents, in this operation there are some
application for agents which will be explained later.
 Outgoing calls: in this section you can determine some numbers;
call center automatically call them and connected them to the
agents. About the application and how to activate will discuss later.
7.5
making agent and its configuration
I‘ll try to explain the menu based on priorities and performance. In
first step we should make an agent. Go to call center menu, agents, click
on New Agent. The page below will be open.
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Agent Number: unique number for agent that can be entered to
the agent console. It is good that your agent number be more than
3 digits (it is not required).
Password: password for agent number.
Retype Password: enter it twice for reducing the risk of confusion.
Name: a name for your agent. It can be good.
After filling out and saving, you will see your agent. It will be offline
because you don‘t enter through console.
It is the important section and cause many cannot install a call
center. In this stage you should enter your defined agent in a predefined
queue in menu of telephony system of PBX.
For this purpose we enter to the queue and write the agent number with
an A character before that in a static agents as follows:
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Your agents are in the queue, but still there is something to set up a
call center with the least setting and it is adding your queue to call
center module.
7.6
ingoing calls
It has two sub-menus which one of them was used to define queues
in call center and the other for entering the names and numbers you
work with (your clients).
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7.6.1
Queues
Queue is a part of a system that manages the ingoing calls and
connected to the agents. In this section you can add the queues that are
defined already in telephony system menu (PBX) & Queue and active or
inactive them.
A. Select queue:
- Select queue: select the queue from the list.
- Script: text played for agent.
B. View the queue:
- Queue list: Queue list that transferred the ingoing calls.
- Queue: name of queue
- Status: being the queue active or inactive
- Options: there are 2 selection: one of them is ―view‖ for viewing
the details and likely changes in queue and the other is ―activate‖
for activating the deactivating queues.
7.6.2
Clients
In this section you can save list of clients and their phone numbers.
In fact you make a phonebook. Call center module use this when
receiving calls to show the details of the caller.
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You can load your list in a CSV file in this format:
―Phone number‖,‖company‖,‖name‖,‖last name‖
7.7
Breaks
In Break you can clear a reason for resting or exiting the agent.
Agents can be free for a while by choosing Breaks for example we define
Breaks for lunch time of an agent.
2parameters is asked for making Breaks:
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time,
resting,
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Description: an explanation about the Breaks
After making Breaks you can active or inactive it in options which is
activated as a default.
7.8
Forms
You can make a form with objects you have. Use of a form is when a
call is connected to an agent, and he can select a form during a
conversation with client and include information in it. For example you
design a form for order product and agent can fill it during the call.
A. Form designer
In this sub menu, by clicking on Create new form you can enter the
necessary information for designing a form. The details are shown here:
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Name: select a name like: selling form
Description: a short explanation about the form such as selling
form of phone card
Name of Field: the name of field which is supposed to be shown.
Order: Show the priority of field in the form
Type: type of your field. You can have different type of field
according to your needs such as:
 Type label: it is used for showing a text such as description or
giving an attention to an agent
 Type text: a field with capability of filling a text, short texts
such as name and last name or name of company
 Type list: it can list your products or anything which an agent
wants to select one of them.
 Type date: date field
 Type text area: it‘s a field for entering the long texts such s
address or description.
A designed form is shown for selling a phone card:
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B. Form preview:
In this sub menu you can see the forms as agent can see. Image
below is the form Preview that we made.
7.9
outgoing calls
A. campaigns:
In this section system calls the lists you have given automatically and
connect it to your queue. They name it campaigns because the most
widely use of this application is group call or campaigns. Another use of
this example can be expressed: campaigns help you to declare an
announcement, warning or new services to your pervious customers.
B. Making campaign:
After clicking on Create New Campaign, a form with required field for
making a campaign will be open.
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Description of the field is as follow:
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Name: a name for this call group such as ― new cards of sangoma‖
Range Date: determining a time-span for calling, start date for
starting the calls and end date so that if the calls didn‘t end until
that date, system wouldn‘t continue the call.
Schedule Per Day: calls for daily schedule
Form: you can choose either form you have made for collecting
information. After connecting to the agent, the selected form will be
shown to insert necessary information.
Trunk: selecting the trunk that you want the system use for
outgoing calls.
Max Used Channels: it is possible that you don‘t want the
campaigns have negative impact on whole lines of the company. So
you try not to busy the lines and consider some restrictions. In this
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field you determine the number of trunk channel that system has
right to use for this purpose.
Context: for this campaign the name of context is Form-Internal
and there is no need to change it.
Queue: determining a queue for accountability. After calling,
system put this numbers into the queue.
Retries: the frequency that system tries to call a number (System
for various reasons may not be able to call numbers such as: being
the line and the number busy, wrong number and the number not
respond…)
Call File: a text file of CSV which includes phone numbers.
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Example: a simple phone file
Note: phone numbers should place in a column. If they have titles, the
first line should start with ―,‖.
Note: first column should always include phone numbers. So if you have
several columns, you don‘t need to write the title of the phone numbers.
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Example: file of customer list with the name and address
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Script: a message for agent when a call is connected.
C. View and change:
The example is as follows:
As you can see you can activate or deactivate this Campaign.
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Console agent is an environment that a made agent can enter with a
number and extension. As soon as the agent entered, he is responsible
for accountability of the calls in queue.
A. Entrance:
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Agent number: it is the same number you made in agent menu.
Extension: an extension number that agent wants to be
accountable.
After pressing Enter, the entered extension starts to ring. After
answering, the system asks your password that you entered in agent
number and console page will be open and you will hear music in your
phone until you connect to the first call. In fact an agent should never
hang up the phone!!!! So agents use headset.
7.10
Console environment:
After entering the password through phone, console of agent will be
open as follow.
(In next version of the book will be completed)
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8. Custom Context Module
This module is a side add-ons of freepbx which is not installed on
Elastix as a default. You should receive it from Freepbx.org or VoIP-IRAN
on sourceforg.net:
http://mirror.freepbx.org/modules/release/contributed_modules/
You can classify the extensions with this module and added
application to freepbx and supervise on their calls separately. This
module is appropriate for those wanted to do have several separated
group work on an Elastix server. Other features of this module are:
classifying the extensions for having different group on a telephony
system, restrict to access some of extensions to other system resources
such as trunks, fax systems and…
For installing this module you should download the link and then enter to
freepbx. Go to menu of ―call center‖,‖telephony system configuration‖
and unembedded freepbx. For entrance use admin user and password (if
you use Elastix version 1.6 or older, the password is admin too).
In freepbx for installing the new module you should go to ―tools‖,
―module admin‖ and click on Upload Module.
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After clicking on Upload Module you can load the received file of Custom
Contexts.
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After loading, in menu of Module Admin and list of Modules and in Third
Party Add-on, custom contexts is added. Select it, click on install and
then process.
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After installing in set up menu, at the end of the list, in third party addon custom contexts is added. You can define new contexts there. In
defining a custom context you may face the following:
Context: select a name. It is used in defining an extension and
determining contexts.
Description: an explanation for contexts.
Dial Rules: if it defines you can run the rules for dial plan. You can use
dial patterns, for example if you put 2xx, it means the entire dialed plan
are 3-digit and start with number 2.
You can see a list of telephony system application that by Allowing them
you let extensions with this contexts use them.
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After defining custom contexts go to extensions. You will see an
extension or changing in defined extension, custom context is added
that you can determine this contexts or this extension. In fact you will
put an extension in a specific group that you defined.
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9. Instant messaging with OpenFire
It is a popular chat program and use Jabber/XMPP protocol for
exchanging data. After installing this program you can have services
such as Google talk, yahoo messenger and etc. name of client program
installed in staff computers is SPARK which they will have these features
by the configuration you did:
-
Chat
Exchanging the file
Calling an extension by pressing a key
By installing add-ons you can add your friends with Yahoo ID, MSN
ID and … in fact by having a proper configuration and internet you
can add your friends in yahoo messenger and chat with them.
9.1
Installing Openfire
After clicking the instant messaging tab (openfire) you will see this
message, because it is not installed on Elastix as a default.
―Click Here‖ and installation progress will start.
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In first stage you should select the language.
In this section you should fill the domain part with the name of Host or
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your IP server which is recommended don‘t change the name system
recognized! You can change the console ports if you want but it is better
to use default ones.
In next stage you should select how you want to be connected to
database. The first one asks you a lot of question about connecting to
the database which waste your time!! Select the second one and
continue.
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In this stage it is asked that where do you want to store users item 2
and 3 is used when you want to store them in a directory server or
clearspace otherwise select Default.
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Determining an email for admin user and select a password for admin
user (this password has nothing to do with your email), try to choose a
password you can remember!! Because its retrieving is very difficult.
Congratulation!! Your openfire installed now by clicking on ―Login to the
admin console‖ go to the page of management.
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Enter with admin user and password you have selected in previous
stage.
Don‘t be sad! You are not supposed to change all the setting! And never
try to update an openfire, this program will be update with any new
versions of Elastix. Updating manually may cause many problems so
don‘t risk.
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Now we go to plugins to install some add-ons (in this stage you need
internet, if you don‘t access, you need to download the add-ons and
upload here).
After clicking on available plugins, lists of add-ones will appear. For
installing the add-ons (Asterisk-IM openfire plugins) click on green sign
(+), after installation, these add-ons will be added to the list of plugins.
Install these add-ons either:
SIP Phone Plugin, Presence Service, and IM gateway
By installing new plugin, Asterisk-IM will be added to the menu. Enter to
in to change some part of it.
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Enable Asterisk-IM and in change the Asterisk queue presence and dropdown device selection to ―Yes‖ and save it.
Asterisk-IM has a bug that should be fixed before using! We should have
small change in file. For convenience you can use Winscp for editing the
files and console environment or nano program. But for changing, open
the file below:
/opt/openfire/plugins/Asterisk-im/database/Asteriskim_hsqldb.sql
You will have something as follows:
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Omit the word unique in third line.
After changing the file should be like that:
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The bug is fixed! Now reboot Elastix to continue.
After rebooting, go to the Asterisk menu, general settings and click on
Add Server and make the necessary adjustments.
-
-
Server name: name of your server which is the Host name. in this
example it is voip-iran.
Server address: IP number 127.0.01
Port: 5038
Username: admin
Password: as a default is ―Elastix 456‖
Configuration should be as follow:
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After clicking on Create Server, the server should be made and shown
with a green ball. If there was nothing after creating server, be sure that
you made a mistake in fixing bugs and rebooting the system. If it was
created but it was shown with a gray ball, so probably registering on
Asterisk is the problem. Once more check your server settings,
especially the password. To ensure that the password, Elastix456, is
correct you can check the correctness in the following file:
/etc/Asterisk/manager.conf
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Now we are going to define a new user, the user that should be defined
on SPARK and registered on the server. Go to users/groups menu. As
you can see there is a default user which is admin.
Click on Create New User.
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Fill the necessary fields. For convenience it is better to enter your
username with small words. You will see following page after filling and
clicking on Create User.
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We have a user but if we want to connect our user with one of the
extensions of telephony system, we should make a map (in fact we want
to consider an extension for this user). go to Asterisks-IM below the
Phone Mappings.
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Username: the user we already made.
-
-
Device: defined extension in telephony system which SIP/ Protocol
is set there. You can use from the extension of the list.
Extension: Intercalate the extension number.
The other 2 items don‘t need any changes. Now click on Add.
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Well we have done here. Now we should install a SPARK IM client and
test it.
9.2
Installing SPARK
As we mentioned earlier, SPARK is an IM client that you can be
registered on openfire and communicate with others exactly like Yahoo
Messenger. Receive the latest version of SPARK from this link and install
it.
http://www.igniterealtime.org/projects/spark/index.jsp
After installation you will see the first page.
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Enter username and password, for server also enters IP of your
telephony system and then login.
At first go to menu contacts and add others with their users. Of course
other side should allow to be added. You can have conference with this
program, exchange the file and more other features. It is a very
exciting service for any company, I hope enjoy.
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10. RoomX Module
10. 1
Configuration
 General
Setting operating mode.
Here, you could select 2 operating mode. (Hotel, and Hospital). Today,
only one operating mode is enable.
 Functions
You could select 3 basic functions in RoomX application.
Locked when checkout. When the room will billed, this room will
locked. So impossible to calling a number.
Calling between rooms: When checked, the room is able to call
another room, but only if this room is included into the same group as
the called room.
Room must be clean: The room appear into the list of available room
only if the room is cleaned. Else, the room will not appear into this list.
However, this room could appear if you need to make a booking about
this room.
Company.
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You can customize your company header, like the logo (png, or jpg file
extension), the company address, and the professional mail of company.
 RoomX Dial Plan
You can customize or change the prefix of each RoomX function. 3 Prefix
exist right now.
Mini-bar, is able to add some drinks on the room, and will used during
the billing. When the chambermaid will cleaned the room, she could
check the mini-bar and enter all drink used by the guest.
Eg: *37122*
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(1 could be a Coca, and 2 could be a Whiskey). So 1 Coca and 2
Whiskey, total 3 drinks.
Room Clean Prefix will used when the room will cleaned by the
chambermaid.
Reception is here to giving a phone number to the reception..
 Tax
Two tax values can be entered. The first value is used by the outbound
calls during the billing.
10. 2
Models
 Model List
Models display all models already recorded into RoomX configuration.
You could delete a room or more just selecting the checkbox at left of
row.
 Add Model
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Before to add any room, you must create some models to putting them
on each room.
Just putting a model with its price, and why not, enter a price to
additional guest if you want, and select the V.A.T used by this room. (2
V.A.T. are enabled).
 Add Rooms
With this module, you can select the rooms that you want from Freepbx,
and into each line, select the type model for room. Don't forget, try to
prepare a good list of name for each room. (eg: room 100, room
101..etc). This name will be used by RoomX if no name is entered.
10. 3
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This menu affecting a product on each key with its price. You can enter
10 different products on this module.
2 V.A.T can be selected.
When the chambermaid check a mini-bar, she dials the prefix (*37 by
default) and presses the number of product used. The entering is ended
by the * key. If the # key is pressed, all products will not billed.
10. 4
Checking
You can making a checking with this menu. You can see some fields to
enter different values.
Room, displays all available rooms into this list, and all cleaned room (if
this options is enabled or not into config parameters).
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Date, it's the Checking date ( the current date by default).
Date Checkout, is needed to have a reference in the case where
another room will booked. This information is not used for the billing.
Of course, you must enter a last-name and first-name to making a
checking.
This first part is required and needed to making a checking. The others
fields below are facultative.
However, only one field is needed in the case where you want to sending
the billing by mail. In this case, you must enter this field. (mail). No
billing will sent by fax yet.
If you want to make a booking, you must click on Booking and why not,
maybe click on
to have the full list of rooms (free and not cleaned).
When no room are into the list, check if all rooms are cleaned and
available. If it's done, then your hotel is full.
10. 5
Room List
You can see the room's status currently in your hotel. The guest name,
the room name, if it's free or busy, cleaned or not. If the guest used the
mini-bar or not. If the room is on DND status or not ( Do Not Disturb ).
And you can see if the room is included in a group or not.
If the phone device is a SIP phone, you can know if the phone is
connected or not. In this case, you have a small yellow triangle beside
the phone number.
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10. 6
Group List.
Here, you can see all group already existing, and you can add lots of
rooms into a group in the same time. Just selecting several rooms
maintaining the [shift] key and click on the rooms that you want.
10. 7
Checkout
You can do 2 types of checkout. A classic checkout by room, and a
checkout by group.
Checkout by group will take all room in group, and will make the
checkout, room by room.
If paid is checked, the billing is paid by the guest, else, this billing is
taken like not paid.
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If you want to have all calls details for the room, check this box.
If you want that guest receive its billing by mail, check this box.
When you click on save button, these information will appear:
If the phone is unconnected, you could see this information:
Else, you will have only:
Next, you will there information below:
Click on display to see the billing, and you could print this billing by the
way.
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10. 8
Billing Report
When a guest will paid its billing, then you could check the paid box.
10. 9
Booking
 Booking Status
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Here, you have all booking which currently entered into RoomX. You can
do a view between 2 dates.
 Booking List
To make a checking on a booked room, check the checking box, and if
you want to cancel a booking, check the canceled box.
 Company report
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You can realize some company report, like how many checking and
checkout by day between two dates.
Sum Rooms, mini-bar, calls, and billings.
(The details will coming soon with the next versions)
10. 10 Home
On the Home page, you can see some information directly on this page.
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How many available rooms (rooms free), or not (rooms busy), if you
have some booking today, the total rooms, if your Hotel is full or
potentially full (caused by the booking).
If there's a booking today, just click on the button to going to booking
menu directly.
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11. Asterisk AGI Programming
11.1
What is AGI?
The Asterisk Gateway Interface, or AGI, provides a standard interface
by which external programs may control the Asterisk dialplan. Usually,
AGI scripts are used to do advanced logic, communicate with relational
databases (such as PostgreSQL or MySQL), and access other external
resources. Turning over control of the dialplan to an external AGI script
enables Asterisk to easily perform tasks that would otherwise be difficult
or impossible.
Asterisk AGI enables an IVR developer to develop IVR structures that
are sometimes, bordering on the absurd, as applications tend to become
more and more complex by using AGI. However, there are some
scenarios where common dialplan practices are no longer applicable, and
the use of an external logic is a must.
11.2
Calling an AGI Script from the Dialplan
In order to work properly, your AGI script must be executable. To use an
AGI script
inside your dialplan simply call the AGI() application, with the name of
the AGI script.
You can do it by adding your code to extensions_custom.conf file in
/etc/asterisk
as the argument, like this:
include => test
[test]
exten => 123,1,Answer()
exten => 123,2,AGI(agi-test.agi)
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AGI scripts often reside in the AGI directory (usually located
in /var/lib/asterisk/agi-bin), but you can specify the complete path
to the AGI script.
11.3
AGI, EAGI, DeadAGI and FastAGI
In addition to the AGI() application, there are several other AGI
applications suited to
different circumstances. While they won‘t be covered in this chapter,
they should be
quite simple to figure out once you understand the basics of AGI
scripting.
The EAGI() (enhanced AGI) application acts just like AGI() but allows
your AGI script
to read the inbound audio stream on file descriptor number three.
The DeadAGI() application is also just like
a channel
, but it works correctly on
AGI()
that is dead (i.e., a channel that has been hung up). As this implies, the
regular AGI()
application doesn‘t work on dead channels.
The FastAGI() application allows the AGI script to be called across the
network, so that
multiple Asterisk servers can call AGI scripts from a central location.
In this chapter, we‘ll first cover the sample agi-test.agi script that comes
with Asterisk
(which was written in Perl), then write a weather report AGI program in
PHP, and finish
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up by writing an AGI program in Python to play a math game.
To get a list of available AGI commands, type show agi at the Asterisk
command-line interface. You can also
*
refer to Appendix C for an AGI command reference.
11.4
AGI scripting frameworks
As with any other open source project, the number of frameworks
built for the development of AGI scripts is amazing. Considering the fact
that the AGI language consists of less than thirty different methods, the
existence of over thirty different scripting frameworks is amazing.
The following list contains some of the more popular frameworks for AGI
scripting:
Language
Framework
URL
PERL
Asterisk PERL
Library
http://asterisk.gnuinter.net/
http://sourceforge.net/projects/phpagi/
PHP
PHP-AGI
Python
py-Asterisk
C
libagiNow
http://py-asterisk.berlios.de/pyasterisk.php
http://www.open-tk.de/libagiNow/
http://gundy.org/asterisk
.NET
MONO-TONE
JAVA
Asterisk-java
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http://www. asterisk-java.org
11.5
The ten rules of AGI development
Rule #1: An AGI script should terminate as fast as possible
First-time AGI developers tend to develop their entire application within
an AGI script. As you develop your entire application within an AGI
script, you may gain the power of the scripting language, but will incur a
cost of performance. Always make sure that the AGI scripts that you
develop terminate their execution as fast as possible, returning to the
dialplan as fast as possible. This concept dictates that each AGI script
being run should behave quickly as an atomic unit—hence the name
"Atomic AGI".
Rule #2: Blocking applications have no place in AGI
As a direct continuation to rule #1, you should never execute a blocking
application from within an AGI script. Initiating a blocking application
from within an AGI script will make your Asterisk environment explode
slowly. Why is that? Because for every blocking application that you run
from within the AGI script, you will have both your AGI script and the
blocking application running for the duration of the block. Imagine that
you were to initiate the Dial application from within an AGI script, and
the call created would last over thirty minutes. For those thirty minutes,
your AGI script is still active. This isn't much of a problem when dealing
with small-scale systems. But when trying to run 50 concurrent scripts,
be prepared for failure.
Blocking applications include the following: Dial, MeetMe, MusicOnHold,
Playback (when dealing with long playbacks), Monitor, ChanSpy, and
other applications that have an unknown execution duration.
Rule #3: Asterisk channels are stateful—use them
An Asterisk channel, once operational, is like a big bucket of information.
Channel variables can be used to carry information from your AGI script
to the dialplan and back. The variables remain as part of the channel
until the channel is terminated, when memory is then freed.
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Using this "bucket" enables you to carry variables and information
obtained via an AGI script and then pass these to an application in the
dialplan. For example, imagine that you are developing a pre-paid
platform. A decision on the call timeout is taken via an AGI script.
However, the actual dialing of the call is performed from the dialplan
itself.
Rule #4: AGI scripts should manipulate data—no more
Most developers tend to think of AGI scripting as a functional unit,
meaning that they enclose multiple functionalities into the AGI script.
While AGI is fully capable of performing telephony functionality, it is best
to leave this functionality to the dialplan.
Utilize your AGI script to set and reset channel variables and
communicate with out-of-band information systems. The concept of
allowing out-of-band information flow into Asterisk's operational flow of
channel, enables new functionality and possibilities. Not all logic should
be handled by your AGI script. Sometimes, it is better to use the AGI
script as a data conduit, while letting an external information system
handle the data manipulation.
Rule #5: VM based languages are bad for AGI scripting
Virtual machine based programming languages are bad for AGI scripting.
Putting aside the rules #1 and #2, imagine that your application is built
using an AGI script using the Java programming language. If you are
familiar with Java, you most probably know that for each program that
you execute using Java, a full Java virtual machine is invoked.
While this practice may seem fairly normal for information systems,
Asterisk and IVR development vary. Imagine that our system is required
to handle a maximum number of 120 concurrent channels, which means
that the maximum number of concurrent AGI scripts will be 120.
According to this concept, our Java environment will be fully invoked for
each of these 120 instances. In other words, too many resources are
being used just for the VM.
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If you really feel that you want to develop AGI scripts using Java,
FastAGI is the way to go for you.
Rule #6: Binary-compiled AGI is not always the answer
While evaluating rules #1, #2 and #5, we can't but reach an almost
immediate conclusion that we need to have our AGI script binary
compiled. While in terms of efficiency and performance, a binary
compiled AGI may provide better performance, the time required to
develop it may be longer. In some cases, scripting languages such as
PHP, PERL, and Python may provide near-similar performance at a lesser
cost.
Also, when using binary compiled AGI scripts, you are always in charge
of the memory allocation and cleanup. Even the most experienced
developer can commit errors while dealing with memory allocation, so
binary compiled AGI need not be the solution always.
If your system truly requires the performance edge of a binary compiled
AGI, we encourage you to develop a prototype using a scripting
language. Once you have your prototype working, start developing your
binary version.
Rule #7: Balance your scripts with dialplan logic
While evaluating rules #1, #2 and #4, we must keep in mind that
developing IVR systems with Asterisk resembles a high-wire balancing
act. The fine art of balancing your dialplan with AGI scripts proves to be
a powerful tool, especially when developing complex IVR systems.
Many developers tend to externalize functionality from the dialplan into
AGI, while the same functionality can be achieved by writing dialplan
macros or dialplan contexts. Using Asterisk's branching commands
(goto, gotoif, exec,, execif, gosub and gosubif) can easily remove
redundant AGI code, passing the control from the AGI back to the
dialplan.
Rule #8: Balance your scripts with web services
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When evaluating rule #4, one may ask: "What is an out-of-band
information system?" We shall explain now. Most Asterisk developers
tend to develop their systems with the data information system—either
embedded into their Asterisk server or communicating with an
information system installed on the same server with the Asterisk
server.
While, for small systems, this proves to be both efficient and economic,
when developing a high-end system that requires scalability and
redundancy, this methodology proves to be counter-productive. One of
the methodologies (although many others exist) for interconnecting
Asterisk with an out-of-band information system is web services.
Communication to the web service is performed via AGI; the webservice protocol—you can use your favorite one.
The choice of protocol isn't that important, as almost any protocol type
used for web services would do. Be it SOAP, WSDL, XML-RPC, WDDX or
any other, take your pick, and the performance and scalability should be
similar in any of these.
Rule #9: Syslog is your friend—use it
Every developer knows that using log files for debugging and monitoring
purposes is a must. Be it for using a binary compiled AGI or a scripting
language based AGI, make sure to utilize the logging facility. Trying to
debug an AGI application from within the Asterisk console, though
possible, can prove to be a tedious task. Sending log entries to a wellformatted log can save you much time and headache.
Scripting languages, such as PHP and PERL, do not offer a direct
debugging facility, making the debugging of such AGI scripts even
harder. Using log files as a debugging point for your AGI script will prove
very useful when developing highly complex systems.
In order to make your syslog more readable, assign a self-created
unique ID to each of your calls. When writing to your log, make sure
that the unique ID appears in each log entry, so that you can trace a
specific session flow through Asterisk. Remember, an Asterisk channel is
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stateful. The unique ID will remain as part of the channel until it is
removed from the system.
Rule #10: The Internet is for Asterisk
As bad as the following may sound, if you have a problem or an idea,
remember that someone else had almost definitely come across it before
you did. I don't want to discourage you, but actually, I want you to
make use of the multitude of Asterisk resources available on the
Internet.
The amount of information relating to Asterisk and platform
development that has been accumulated by search engines is
staggering. Over the course of the past two years, the amount of
information available has multiplied two times (at least), making it the
best source to find answers to your questions.
Asterisk user forums exist today in almost every country around the
world; in some countries, there is more than one forum. These forums
provide fast answers and professional guidance, allowing you to
concentrate on your development, instead of concentrating on obtaining
information.
11.6
AGI Commands
* answer: Asserts answer
* asyncagi break: Break Async AGI loop (since Asterisk 1.6)
* channel status: Returns status of the connected channel
* control stream file: Send the given file, allowing playback to be
controlled by the given digits, if any. (since Asterisk 1.2)
* database del: Removes database key/value
* database deltree: Removes database keytree/value
* database get: Gets database value
* database put: Adds/updates database value
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* exec: Executes a given Application. (Applications are the functions you
use to create a dial plan in extensions.conf ).
* get data: Gets data on a channel
* get full variable: Gets a channel variable, but understands
complex variable names and built-in variables. (since Asterisk 1.2)
* get option: Behaves similar to STREAM FILE but used with a timeout
option. (since Asterisk 1.2)
* get variable: Gets a channel variable
* hangup: Hangup the current channel
* noop: Does nothing
* receive char: Receives one character from channels supporting it
* receive text: Receives text from channels supporting it
* record file: Records to a given file
* say alpha: Says a given character string (since Asterisk 1.2)
* say date: Say a date (since Asterisk 1.2)
* say datetime: Say a formatted date and time (since Asterisk 1.2)
* say digits: Says a given digit string
* say number: Says a given number
* say phonetic: Say the given character string.
* say time: Say a time
* send image: Sends images to channels supporting it
* send text: Sends text to channels supporting it
* set autohangup: Autohangup channel in some time
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* set callerid: Sets callerid for the current channel
* set context: Sets channel context
* set extension: Changes channel extension
* set music: Enable/Disable Music on hold generator, example "SET
MUSIC ON default"
* set priority: Prioritizes the channel
* set variable: Sets a channel variable
* speech activate grammar: Activates a grammar (since Asterisk 1.6)
* speech create: Creates a speech object (since Asterisk 1.6)
* speech deactivate grammar: Deactivates a grammar (since Asterisk
1.6)
* speech destroy: Destroys a speech object (since Asterisk 1.6)
* speech load grammar: Loads a grammar (since Asterisk 1.6)
* speech recognize: Recognizes speech (since Asterisk 1.6)
* speech set: Sets a speech engine setting (since Asterisk 1.6)
* speech unload grammar: Unloads a grammar (since Asterisk 1.6)
* stream file: Sends audio file on channel
* tdd mode: Activates TDD mode on channels supporting it, to enable
communication with TDDs.
* verbose: Logs a message to the asterisk verbose log
* wait for digit: Waits for a digit to be pressed
Note: When Asterisk starts an AGI script, it feeds the channel variables
to the script on standard input. The variable names are prefixed with
"agi_" and are separated from their values by a colon and a space.
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Though the actual channel variables may be in the upper case, the
names passed to an AGI script are all lower case. Global variables are
not passed to the AGI script in this manner.
* agi_request - The filename of your script
* agi_channel - The originating channel (your phone)
* agi_language - The language code (e.g. "en")
* agi_type - The originating channel type (e.g. "SIP" or "ZAP")
* agi_uniqueid - A unique ID for the call
* agi_version - The version of Asterisk (since Asterisk 1.6)
* agi_callerid - The caller ID number (or "unknown")
* agi_calleridname - The caller ID name (or "unknown")
* agi_callingpres - The presentation for the callerid in a ZAP channel
* agi_callingani2 - The number which is defined in ANI2 see Asterisk
Detailed Variable List (only for PRI Channels)
* agi_callington - The type of number used in PRI Channels see
Asterisk Detailed Variable List
* agi_callingtns - An optional 4 digit number (Transit Network
Selector) used in PRI Channels see Asterisk Detailed Variable List
* agi_dnid - The dialed number id (or "unknown")
* agi_rdnis - The referring DNIS number (or "unknown")
* agi_context - Origin context in extensions.conf
* agi_extension - The called number
* agi_priority - The priority it was executed as in the dial plan
* agi_enhanced - The flag value is 1.0 if started as an EAGI script, 0.0
otherwise
* agi_accountcode - Account code of the origin channel
* agi_threadid - Thread ID of the AGI script (since Asterisk 1.6)
Using the "get variable" AGI command to get values above.
11.7
simple PHPAGI example
at first make extension 888 in extension_custom.conf like this
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include => test
[test]
exten => 85,1,Answer
exten => 85,n,Wait(2)
exten => 85,n,AGI(test-php.php)
exten = 85,n,hangup()
secondly make your test.php file in this path /var/lib/asterisk/agi-bin
and don‘t forget to give it execution permission.
#!/usr/bin/php -q
<?
set_time_limit(30);
require('include/phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi->answer();
$agi->stream_file("demo-congrats","#");
do
{
$agi->stream_file("enter-some-digits","#");
$result = $agi->get_data('beep', 3000, 20);
$keys = $result['result'];
$agi->stream_file("you-entered","#");
$agi->say_digits($keys);
} while($keys != '111');
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$agi->hangup();
?>
Now reload asterisk and call 888 from your extension.
11.8
Interact with MySQL inside PHP-AGI
The following AGI retrieves asterisk extensions from Asterisk database in
MySQL and displays them as verbose.
#!/usr/bin/php5
<?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
mysql_connect('localhost','root','SmartPBX');
$agi -> verbose('connection set');
mysql_select_db('asterisk');
$sql = "select extension, name from users";
$agi -> verbose($sql);
$result = mysql_query($sql);
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while($row = mysql_fetch_assoc($result))
{
$rows = "Ext # :{$row['extension']} " . " *** | *** " .
"Name : {$row['name']}";
$agi->verbose($rows);
}
mysql_close();
?>
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12. Tips and tricks
12.1
connecting 2 servers of Elastix together
It is possible to connect two servers with different ways, the easiest and
most reliable way is connecting with IAX protocol. The following will
express how to define this relationship. The first step in the trunk of
IAX2 is drawing a picture from whatever you need. Here we have a
simple connection between two PBX, for convenience name them 101
and 106 (according the IP of two PBX).
All the extensions of PBX 101 start with 1xxx and all the extensions of
PBX 106 start with 3xxx.
This is good for creating outbound route.
Trunks of IAX2 are named according to their destinations and are shown
with arrows.
There is information for configuration in two sides of PBXs, as it is shown
in PBX 101 it is needed to have outbound trunk which is peer-106 and
user-106. Similarly, in PBX 106 some information for configuration is
needed for having an outbound trunk with peer-101 and user-101.
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We start by having a trunk when both PBX have IAX2 trunk, we define
outbound route.
 Setting trunks of IAX2
IAX2 trunk in PBX101 & PBX106
A. Select add trunk in main menu of FreePBX
B. Select add LAX2 Trunk
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C. Don‘t change anything, in Outgoing Settings for configuration of
PBX 106 and PBX 101 do as follow:
ELASTIX-SERVER1(192.168.1.101)
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ELASTIX-SERVER1(192.168.1.106)
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D. Outgoing setting will be configured by basic information that is
destination of IAX user.
Configure outgoing settings and incoming settings in PBX 101.
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Outgoing Settings:
*Trunk Name: 106-IAX it can be any name.
*192.168.1.106:host_
destination trunk.
*username=101-IAXuser
PBX.
it is the name of domain or address of
it is LAX2 user which is made in destination
*secret=101-IAXpassword password used for confirming the trunk
connection in destination PBX.
*type=peer kind of LAX2 connection. This connection is from one PBX
to another PBX.
*qualify=yes it register connection with the destination.
Incoming Settings:
*user context: 106-IAXuser it can be any name.
*secret=101IAXpassword
password used for confirming the trunk
connection in destination PBX.
*type=user kind of IAX2 connection. Peer is confirmed by user account.
For configuring the outgoing settings and incoming settings in PBX106
follow the configuration in image.
E. Click on submit, update and reload it. And check it with webmin or
other management softwares. File of / etc/Asterisk/iaxadditional.conf in PBX101 should be configured as follow:
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And file of /etc/Asterisk/iax-additional.conf in PBX106 as below:
F. Go to the next level and test the trunks of IAX2
 Testing the trunks of LAX2:
Go to Asterisk console for testing the trunks (CLI), you can do it with
several ways.
 Through console of PBX
 Through doing SSH by using Putty SSh
 Through FreePBX/Webmin SSH
Enter the following command in Linux.
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root@pbx~$ Asterisk –rvv
(One r & two v)
In Asterisk click on command of iax2 show peers.
Above image shows the connections of PBX 101 IAX peer. Pay attention
to its important parts:
 Above image shows that peer-106 is connected by user-101
 Status should be ok, change it if it was something else.
 Client of 1002 and IAX2 are connected and are 2 high peers.
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Above image shows the connection of PBX 106, IAX2 peer. The
important parts are:
Above Image shows that peer-101 is connected by user-106
 Status is ok, check the setting if it was something else.
 Client of 3002 of IAX is connected.
 And 2 peers are online.
Next stage is definition of outbound route for PBX 101 & PBX 106.
 Setting outbound routes
There are 2 options for setting the outbound routs: Direct call to the
extension or entering number 7. Both will be described.
Outbound routs in PBX 101
There are 2 ways for outbound routes, select one of them and start with
outbound route in the menu.
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First_ calling directly to the extension, PBX101&PBX106
Rules of outbound allow PBX 101 call the extensions of PBX 106 directly.
For example for calling to extension 3001, dial 3001,PBX 106.and also
rules of outbound allows pbx 106 to call directly to extensions of
PBX101. For example for calling to extension 1001, dial 1001,pbx 101.
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Configure these in PBX 101:
*route name: 106-dial7ext_ it can be anything
**|7_ shows that everything after 7 will be forward to destination trunk.
*trunk sequence: IAX2/106-peer_ this trunk is made for going to the
PBX 106.
Configure these in PBX 106:
*route name: 101-dial7ext: can be anything.
**|7_ shows that everything after 7 will be forwarded to destination
trunk.
*trunk sequence: IAX2/101-peer_ this trunk is made for going to the
PBX 101
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You should send IAX or POST to peers of each PBX from any SIP
extension.
You can check status of online phones, trunks through FreePBX window.
In making trunk, there is no limitation in number of channels that can
use trunk. In above window, 6 channels were connected in 3 connections
through a trunk of IAX2. Three IP phones, two IAX2 s100i POSTS
attached to adaptors of IAX2 and a FXS pots phone.
12.2
Video call on Elastix
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For having video call on Elastix you should do following configuration. Go
to this link. Open the file of ―sip_general_additional.conf‖ for edit. If
there was no file, make it: ―vi /etc/Asterisk/sip_general_additional.conf‖
Add this configuration:
Videosupport=yes
Maxcallbitrate=384
Allow=h261
Allow=h263
Allow=h263p
Allow=h264
And reload it with this command:
Asterisk -rx "module reload‖
in freePBX version 2.7 you can set it easily from graphical environment.
So go to Elastix menu, PBX configuration and click on sub menu of
Unembedded FreePBX to enter freepbx program. Then go to Tools, sub
menu of Asterisk SIP setting or Asterisk IXA setting and enable the video
support.
Some of softphones that supports video are listed:
Bear in mind that you should activate video support on any softphone
you use.
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software
url
platform
Ekiga
http://www.ekiga.org
Linux
Adore
Video
http://www.adoresoftphone.com/softphones/soft
phone-video.html
Window
s
Eyebeam
http://www.counterpath.com/index.php?menu=
Products&amp;smenu=eyeBeam
Window
s
Bria 2.0
http://www.counterpath.com/index.php?menu=
Products&amp;smenu=bria
Window
s
12.3
Limiting the conversation time
A way for limiting and categorizing the extensions is using customcontext module which will be explained later. But it‘s using has trouble
because of the complexity. Now if you want to limit time of conversation
for all the user you can use this trick. Go to PBX, PBX configuration,
General setting and add L(3600000:300000) in Asterisk outbound dial
command options.
Your configuration will be like this:
trL (3600000:300000)
The above command is L (x:y). x is the time allowed for conversation in
milliseconds and call will be terminated in this time and y is remaining
time of the call for announcing the warning for disconnecting it. In fact
this example means that a call will be terminated in 60 minutes and
5minute remaining to the end of the call, a warning announced. For
more information see the following reference.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
12.4
put your Asterisk server behind NAT
 Manually: it is possible that you want to enter your extensions on
internet and be registered on your server. Maybe you don‘t have
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valid IP and put your server behind NAT and you should have
access to internet through that sever. In this case you should have
some configuration on your Asterisk to operate properly behind
NAT and with valid IP, extensions registered on it and contact
establish correctly.
Put these items in ―sip_nat.conf‖ in ―etc/Asterisk‖ branch:
nat=yes
externip=<your fixed external IP> or
externhost=<myAsterisk.freedns.com>
localnet=192.168.1.0/255.255.255.0
externrefresh=10
Extern ip is your valid ip and is map to you. If you use ADSL, it is
your modem IP. In addition, you should forward these ports to the
ip server on your ADSL modem or router.
tcp: 5060
udp: 10000-20000
 Graphical: these can be done through SIP setting in freepbx.
These features is set on freepbx from version 2.7 so go to the pbx
configuration from Elastix menu and click on unembedded freepbx
to enter the freepbx program. Then go to Tools and sub menu of
Asterisk SIP Setting and you will see this page:
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All the parameters that were written in this file can be easily entered
here. Don‘t forget there are always an easy way!!!!
12.5
Installing Codecs of g729 & g723
Every call that is established use a protocol to talk and a compactor
codec for compressed audio packets. It will use less bandwidth if the
packages become more compressed by codec but you will have more
processing load on the processor and the quality of voice will be less.
cd /usr/src
Service Asterisk restart
wget
http://Asterisk.hosting.lv/bin/codec
pentium4.so
g723-ast
14-
icc-glibc-
mv codec_g723-ast14-icc-glibc-pentium4.so/usr/lib/Asterisk/modules
chmod
+x
pentium4.so
/usr/lib/Asterisk/modules/codec_g723-ast14-icc-glibc-
wget
http://Asterisk.hosting.lv/bin/codec
pentium4.so
g729-ast
14-
icc-glibc-
mv codec_g729-ast14-icc-glibc-pentium4.so/usr/lib/Asterisk/modules
chmod
+x
pentium4.so
/usr/lib/Asterisk/modules/codec_g729-ast14-icc-glibc-
Asterisk -rx "module load codec_g729-ast14-icc-glibc-pentium4.so"
Asterisk -rx "module load codec_g723-ast14-icc-glibc-pentium4.so"
Service Asterisk restart
Sleep 2
Asterisk -rx "core show codecs
After installing the packages, you should allow the VoIP protocols use
these codecs. Add ―Allow = g729‖ & ―Allow = g723‖ to
―/etc/Asterisk/sip_custom.conf ―&‖ /etc/Asterisk/sip_custom.conf‖.
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You can add these from graphical environment. For this go to Elastix
menu, PBX configuration and click on Unembeded FreePBX and entered
to FreePBX program. Then go to Tools, sub menu of Asterisk sip setting
or Asterisk IXA setting. You will see following page:
12.6
Asterisk command-line interface (CLI)
Asterisk has an environment that can be changed by some commands or
be a suitable device for controlling Asterisk. For accessing to this
environment which is called CLI, you should enter this command after
entering to Linux.
Asterisk-rvvv
―r‖ means performing the environment and ―v‖ means view details of the
system. The more the number of ―v‖, the more you can see details from
log of Asterisk, but for normal items, 3 ―v‖ is enough.
In this environment you can see whatever Asterisk does like a log. In
addition, you can configure by using commands or view more details.
For seeing the list, enter the ―Help‖ on CLI environment.
This allows you to run a command as if it was typed into the asterisk
CLI. Examples:
* sip show peers
o This displays all the known SIP devices, and their state, according to
Asterisk
* show channels
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o Show any channels that are in use at the moment
* soft hangup Zap/1
o Hangs up the Zap/1 channel
------------------------------------------------------------------General commands
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
debug channel: Enable debugging on a channel
don't 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
remove 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. (:exclaim:) 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
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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 hints: Show registered hints
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
soft hangup: Request a hangup on a given channel
show voicemail users: List defined voicemail boxes
show voicemail zones: List zone message formats
Server management
restart gracefully: Restart Asterisk gracefully, i.e. stop receiving new
calls and restart at empty call volume
restart now: Restart Asterisk immediately
restart when convenient: Restart Asterisk at empty call volume
Note: Restart is more like a reload, not a real restart it just runs the
reload routines (thus open ports are not closed). Often you don't need to
really restart asterisk, instead just e.g. 'unload chan_sip.so' and 'load
chan_sip.so'.
reload: Reload configuration
stop gracefully: Gracefully shut down Asterisk, i.e. stop receiving new
calls and shut down at empty call volume
stop now: Shut down Asterisk immediately
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stop when convenient: Shut down Asterisk at empty call volume
extensions reload: Reload extensions and only extensions
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
AGI commands
show agi: Show AGI commands or specific help
dump agihtml: Dumps a list of agi command in html format
Database handling commands
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
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
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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
H323 channel commands
h.323 debug: Enable chan_h323 debug
h.323 gk cycle: Manually re-register with the Gatekeper
h.323 hangup: Manually try to hang up a call
h.323 no debug: Disable chan_h323 debug
h.323 no trace: Disable H.323 Stack Tracing
h.323 show codecs: Show enabled codecs
h.323 show tokens: Manually try to hang up a call
h.323 trace: Enable H.323 Stack Tracing
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 (clients that register to your
Asterisk server)
sip show registry: Show SIP registration status (when Asterisk
registers as a client to a SIP Proxy)
sip show users: Show defined SIP users
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Zap channel commands
soft hangup Zap/1: Hangs up the Zap/1 channel
zap destroy channel: Destroy a channel
zap show channels: Show active zapata channels
zap show channel: Show information on a channel
MGCP channel commands
mgcp audit endpoint: Audit specified MGCP endpoint
mgcp debug: Enable MGCP debugging
mgcp no debug: Disable MGCP debugging
mgcp show endpoints: Show defined MGCP endpoints
skinny
skinny
skinny
skinny
channel commands
debug: Enable Skinny debugging
no debug: Disable Skinny debugging
show lines: Show defined Skinny lines per device
CAPI channel commands
capi debug: Enable CAPI debugging
capi no debug: Disable CAPI debugging
capi info: Show CAPI info
Sirrix ISDN channel commands
srx reload: Reload channel driver configuration; active calls are not
terminated!
srx show ccmsgs: Disable / enable output of incoming callcontrol
messages.
srx show chans: Show info about B-Channels
srx show globals: Show info about global settings
srx show groups: Show info about configured groups
srx show layers: Show info about ISDN stack (Layer 1, 2, 3)
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srx show sxpvts: Show private info about active channels
srx show timers: Show info about running timers
vISDN ISDN channel commands (Driver which supports Euro ISDN
for HFC chipsets)
visdn reload: Reloads vISDN configuration from the
/etc/asterisk/visdn.conf file
show visdn calls: Shows active calls going through a vISDN channel.
show visdn huntgroups: Shows vISDN huntgroup information
configured correctly in visdn.conf.
show visdn interfaces: Shows configured and available vISDN
interfaces.
debug visdn generic: Enables generic vISDN debugging
debug visdn q921: Enables q.921 debugging
debug visdn q931: Enables q.931 debugging
no debug visdn generic: Disables generic vISDN debugging
no debug visdn q921: Disables q.921 debugging
no debug visdn q931: Disables q.931 debugging
12.7
Asterisk feature codes
There are some codes on Asterisk as a default. Each of them has a
feature and you can dial them as extensions:
## - Transfer (during conversation)
70 - Park Extension
*30 - Blacklist a Number
*32 - Blacklist the Last Caller
*31 - Remove a Blacklisted Number
*72 - Call Forward All Activate
*73 - Call Forward All Deactivate
*74 - Call Forward All Prompting Deactivate
*90 - Call Forward Busy Activate
*91 - Call Forward Busy Deactivate
*92 - Call Forward Busy Prompting Deactivate
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*52
*53
*70
*71
-
Call
Call
Call
Call
Forward No Answer/Unavailable Activate
Forward No Answer/Unavailable Deactivate
Waiting Activate
Waiting Deactivate
** - Call Pickup (dialing another extension)
555 – ChanSpy ( hearing all the channels)
7777 - Simulate Incoming Call
666 - Simulate Incoming FAX Call
*12 - User Logoff
*11 - User Logon
888 – ZapBarge (hearing urban lines)
*35 - Email completed dictation
*34 - Perfom Dictation
*78 - Do Not Disturb Activate
*79 - Do Not Disturb Deactivate
*422 - Connect to Gabcast
*69 - Call Trace
# - Directory
*43 - Echo Test
*65 - Get Your Extension
*60 - Get the Time
*80 - Intercom Prefix
*54 - User Intercom Allow
*55 - User Intercom Disallow
411 - Phonebook dial-by-name directory
*99 - Check Recording
*77 - Save Recording
*75 - Set user speed dial
*0 - Speeddial prefix
*98 - Dial Voicemail
*97 - My Voicemail
12.8
Reading asterisk log files
Perhaps the most important skill required to troubleshoot is reading the
log files. Like other Linux services, asterisk writes the logs in
/var/log/asterisk
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To enable a full log you should edit the /etc/asterisk/logger.conf and
uncomment the full line. Afterwards you must run this command to
apply changes:
# asterisk -rx ―logger reload‖
Here are examples of tracing in logs:
cat /var/log/asterisk/full | grep ERROR
cat /var/log/asterisk/full.1 | grep 9878 | grep “some text”
cat /var/log/asterisk/full|grep “user-callerid: “|grep 9899 >
/var/log/asterisk/9899.log
cat /var/log/asterisk/full.1| grep “[22435]” > 9864.log
tail /var/log/asterisk/full | tee log_exmpl.txt
12.9
Asterisk integration with Cisco Call Manager
Why integrate Cisco CallManager and Asterisk?


Features: Asterisk provides features that CallManager by itself does
not.
Migration: Allow a gradual migration from a closed source PBX to
open source PBX.
There are two ways to accomplish this:
1. Using H.323: In CCM Asterisk appears as a H.323 Gateway.
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2. Using SIP (only in CCM 4.X+)
Here is instruction on how to integrate asterisk with Cisco Call Manager
using SIP trunk.
Step1) On your freePBX create a new SIP trunk and set the following
(remain the rest as is):
host=10.200.214.10
port=5063
fromdomain=10.200.214.10
fromuser=
username=
secret=
type=peer
qualify=yes
canreinvite=no
nat=no
insecure=port,invite
context=from-internal
disallow=all
allow=ulaw
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Step2) you should define your outgoing calls to cisco
Step3)
1. Open up the CallManager Administration web page.
2. Since a SIP trunk requires MTP, make sure you have one:
1. Service -> Media Resource -> Media Termination Point
2. Normally your CallManager server should appear there if you
do an empty query
3. if not, go to the CallManager Serviceabilty web page, and
activate the Cisco IP Voice Media Streaming App service
3. Select Device->Trunk from the menu.
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Select the ―Add a New Trunk‖ link from the upper right hand corner of
the ―Find and List Trunks‖ page.
Select ―SIP Trunk‖ as the ―Trunk type‖ and ―SIP‖ as the ―Device
Protocol‖. Click on the ―Next‖ butto
1. Enter a name in the ―Device Name‖. Valid characters are letters,
numbers, dashes, dots (periods), and underscores. The device
name is only used internally in Call Manager so it can be anything
you want.
2. Enter a description in the ―Description‖ field.
3. Select a device pool.
4. Enter the IP address of your Asterisk server in the ―Destination
Address‖ field.
5. Select ―UDP‖ as the ―Outgoing Transport Type‖.
6. Modify any other settings as needed for your
(CiscoCallManager|CallManager) installation.
7. Click on the ―Insert‖ button.
8. Add route patterns in CallManager that send calls to Asterisk using
the SIP trunk that you just created.
Cisco Call Manager 6.1
It‘ very similar to 4.1, but you must change the UDP protocol of sip in
this menu:
System > Security Profile > SIP Trunk Security Profile
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Outgoing Transport Type: UDP
Source: http://www.voipinfo.org/wiki/view/Asterisk+Cisco+CallManager+Integration
12.10 Customized chanspy
Here is a customized chanspy. It first asks for the password and then
the extension number you want to monitor.
Add this custom extension in /etc/asterisk/extension_custom.conf
[from-internal-custom]
exten => 1234,1,Playback(demo-congrats)
dial 1234
exten => 1234,2,Hangup()
exten => h,1,Hangup()
include => agentlogin
include => conferences
include => calendar-event
include => weather-wakeup
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; extensions can
include => chanspy
[chanspy]
exten = 556,1,Authenticate(1234)
exten = 556,2,Read(SPYNUM,extension)
exten = 556,3,ChanSpy(SIP/${SPYNUM},q)
; end of [app-chanspy]
12.11 Tips and tricks of reading log files
Perhaps the most important skill required to troubleshoot is reading the
log files. Like other Linux services, asterisk writes the logs in
/var/log/asterisk
To enable a full log you should edit the /etc/asterisk/logger.conf and
uncomment the full line. Afterwards you must run this command to
apply changes:
# asterisk -rx ―logger reload‖
Here are examples of tracing in logs:
cat /var/log/asterisk/full | grep ERROR
cat /var/log/asterisk/full.1 | grep 9878 | grep “some text”
cat /var/log/asterisk/full|grep “user-callerid: “|grep 9899 >
/var/log/asterisk/9899.log
cat /var/log/asterisk/full.1| grep “[22435]” > 9864.log
tail /var/log/asterisk/full | tee log_exmpl.txt
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13. Troubleshooting and Maintenance
3.1
Heavy Asterisk Log
Over time Asterisk log can become so heavy and space consuming, not
to mention the time you have to spend reading through tons of lines of
log files. The way to organize the log files is to rotate them using
logrotate facility.
Asterisk has 5 different main log files:
1. /var/logs/asterisk/messages
2. /var/logs/asterisk/queue_log
3. /var/logs/asterisk/event_log
4. /var/logs/asterisk/debug
5. /var/logs/asterisk/full
Here are some parameters description:







missingok: If the log file is missing, go on to the next log file
without issuing an error message.
copytruncate: Truncate the original log file to zero size in place
after creating a copy, instead of moving the old log file and
optionally creating a new one
rotate 7: Log files are rotated 7 times before being removed or
mailed to the address specified in a mail directive. If count is 0, old
versions are removed rather then rotated.
compress: Old versions of log files are compressed with gzip to
save disk space.
size: Rotate only if the log file reaches the specified limit
notifempty: Do not rotate the log if it is empty
sharedscripts
postrotate
/etc/init.d/lighttpd reload
endscript: The lines between postrotate and endscript (both of
which must appear on lines by themselves) are executed after the
log file is rotated. These directives may only appear inside a log file
definition. In our case we are reloading lighttpd. Other opting could
be send HUP single using kill command.
There are different ways you can configure the rotation:
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 Sample 1
/var/log/asterisk/debug /var/log/asterisk/event_log
/var/log/asterisk
/messages {
weekly
missingok
rotate 9
size 2000k
copytruncate
endscript
}
 Sample 2
/var/log/asterisk/cdr-csv/Master.csv /var/log/asterisk/debug
/var/
log/asterisk/event_log /var/log/asterisk/messages {
weekly
missingok
rotate 4
sharedscripts
postrotate
/usr/sbin/invoke-rc.d asterisk logger-reload
endscript
}
 Sample 3
# cat /etc/logrotate.d/asterisk
/var/log/asterisk/queue_log {
missingok
rotate 7
notifempty
daily
create 0640 asterisk asterisk
postrotate
/usr/sbin/asterisk -rx 'logger reload' > /dev/null 2>&1
endscript
}
/var/log/asterisk/event_log {
missingok
rotate 7
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notifempty
daily
create 0640 asterisk asterisk
postrotate
/usr/sbin/asterisk -rx 'logger reload' > /dev/null 2>&1
endscript
}
/var/log/asterisk/full {
missingok
rotate 7
notifempty
daily
create 0640 asterisk asterisk
postrotate
/usr/sbin/asterisk -rx 'logger reload' > /dev/null 2>&1
endscript
}
/var/log/asterisk/messages {
missingok
rotate 7
notifempty
daily
create 0640 asterisk asterisk
postrotate
/usr/sbin/asterisk -rx 'logger reload' > /dev/null 2>&1
endscript
}
3.2
How to tackle Jitter issue
In an environment with old LAN cabling, chances are so high that call
get interrupted, words drop every now and then, and at the worst case,
the sound quality gets terribly unbearable. This is mainly caused by the
issue commonly referred to as Jitter.
In voice over IP (VoIP), jitter is the variation in the time between
packets arriving, caused by network congestion, timing drift, or route
changes. A jitter buffer can be used to handle jitter.
But what exactly does jitter buffer do?
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It trades a slight bit of latency to compensate for jitter, allowing a sort
amount of time for late packets to be added into the stream that would
otherwise have been dropped.
To fix the jitter issue, we must first get to know where we have jitter,
that is, SIP <-> SIP or SIP <-> PSTN
In case of SIP <-> SIP do the following:
1) edit /etc/asterisk/sip_general_custom.conf
2) add the following:
jbenable=yes
jbforce=yes
jbmaxsize=20
3) reload SIP in asterisk:
ippbx*CLI> sip reload
Here are more parameters involved with jitter buffer:
;Enable Jitter Settings
jitterbuffers=4
jbenable=yes
jbforce=no
jbimpl=fixed
;jbimpl=adaptive
jbmaxsize=500
jbresyncthreshold=100
jblog=yes
;tos=0x18
3.3
Echo on POTS lines
Using fxotune will help echo canceller to work better.
Do this before setting up echo canceller, it means to disable echo
canceller options in /etc/dahdi/system.conf and
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/etc/asterisk/chan_dahdi.conf, and to stop asterisk but dahdi still
running.
follow the following step by step:
1. Stop asterisk: /etc/init.d/amportal stop
2. Stop dahdi: /etc/init.d/dahdi stop
3. Remove echo related options to disable echo canceller:
3.1. edit /etc/dahdi/system.conf and set echocanceller=none for all
channels
by default it‘s like this: echocanceller=oslec,1
change to this:
fxsks=1
echocanceller=none
fxsks=2
echocanceller=none
…
3.2. edit /etc/asterisk/chan_dahdi.conf and set the echo parameters like
this:
;echotraining=800 ; disabled when using OSLEC
echocancel=yes ; yes=128. you can use any power of 2 such as
128,256,512,1024
echocancelwhenbridged=no
4. Start dahdi: /etc/init.d/dahdi start
5. Plug the line to FXO
6. To get echoes information on FXO lines run # fxotune -i 4 (use 4, its
any number, but 4 is on all over documents)
7. Once done (2-3 minutes per line/per FXO port), take a look for
/etc/fxotune.conf and read the content, it should have some sort of
configurations or options
8. To apply tuning run: # fxotune -s (this will be automatically
executed by dahdi init, so you dont need to do this everytime)
9. Enable echo canceller options (do not enable on dahdi if you‘re using
hardware echo canceller, only echocancel=yes)
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10. Run asterisk, normal execution, reboot the server or just run
/etc/init.d/amportal start
3.4
Essential Linux commands
Here is a list of some must-know commands in linux with examples:
 grep Command
Option -v, will display all the lines except the match. In the example
below,
it displays all the records from /etc/password that doesn‘t match John.
Note: There are several lines in the /etc/password that doesn‘t contain
the
word John. Only the first line of the output is shown below.
# grep -v John /etc/passwd
jbourne:x:1084:1084:Jason Bourne:/home/jbourne:/bin/bash
Option -c shows the number of lines that the pattern matches.
# grep -c John /etc/passwd
2
You can also get the total number of lines that did not match the specific
pattern by passing option -cv.
# grep -cv John /etc/passwd
39
Use option -r (recursive) to search all subdirectories for a text matching
a specific pattern. In the example below, it will search for the text ―John‖
by ignoring the case inside all the subdirectories under /home/users.
This will display the output in the format of ―filename: line that matching
the
pattern‖. You can also pass the option -l, which will display only the
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name of
the file that matches the pattern.
# grep -ri john /home/users
/home/users/subdir1/letter.txt:John, Thanks for your
contribution.
/home/users/name_list.txt:John Smith
www.thegeekstuff.com
/home/users/name_list.txt:John Doe
# grep -ril john /root
/home/users/subdir1/letter.txt
/home/users/name_list.txt
 find Command
The following command looks for all the files under /etc directory with
mail
in the filename.
# find /etc -name "*mail*"
The following command will list all the files in the system greater than
100MB.
# find / -type f -size +100M
The following command will list all the files that were modified more
than 60
days ago under the current directory.
# find . -mtime +60
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The following command will list all the files that were modified in the last
two days under the current directory.
# find . –mtime -2
How to delete all the archive files with extension *.tar.gz and
greater than 100MB?
Please be careful while executing the following command as you don‘t
want
to delete the files by mistake. The best practice is to execute the same
command with ls –l to make sure you know which files will get deleted
when
you execute the command with rm.
# find / -type f -name *.tar.gz -size +100M -exec ls -l {} ;
# find / -type f -name *.tar.gz -size +100M -exec rm -f {} ;
The following command finds all the files not modified in the last 60 days
under /home/jsmith directory and creates an archive files under /tmp in
the
format of ddmmyyyy_archive.tar.
# find /home/jsmith -type f -mtime +60 | xargs tar -cvf
/tmp/`date '+%d%m%Y'_archive.tar`
 xargs Command
xargs is a very powerful command that takes output of a command and
pass it
as argument of another command. Following are some practical
examples on
how to use xargs effectively.
a. When you are trying to delete too many files using rm, you may get
error
message: /bin/rm Argument list too long – Linux. Use xargs to avoid this
problem.
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# find ~ -name „*.log‟ -print0 | xargs -0 rm -f
b. Get a list of all the *.conf file under /etc/. There are different ways to
get
the same result. Following example is only to demonstrate the use of
xargs.
The output of the find command in this example is passed to the ls –l
one by
one using xargs.
# find /etc -name "*.conf" | xargs ls –l
c. If you have a file with list of URLs that you would like to download,
you can
use xargs as shown below.
# cat url-list.txt | xargs wget –c
d. Find out all the jpg images and archive it.
# find / -name *.jpg -type f -print | xargs tar -cvzf
images.tar.gz
e. Copy all the images to an external hard-drive.
# ls *.jpg | xargs -n1 -i cp {} /external-harddrive/directory
3.5
job scheduling in linux using crontab
I think if you wanna be a Elastix Professional you should know Linux
abilities as best, one of these abilities is cron job.
crontab allows tasks to be automatically run in the background at
regular intervals. You could also use it to automatically create backups,
synchronize files, schedule updates, and much more.
run :
sudo crontab -e
This will open the crontab using the default editor.
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Here is the format to add cronjobs:
* * * * * /bin/execute/this/script.sh
As you can see there are 5 stars. The stars represent different date parts
in the following order:
minute (from 0 to 59)
hour (from 0 to 23)
day of month (from 1 to 31)
month (from 1 to 12)
day of week (from 0 to 6) (0=Sunday)
 example1
So if we want to schedule the script to run at 1AM every Friday, we
would need the following cronjob:
0 1 * * 5 /bin/execute/this/script.sh
 example2
if we want to schedule the script to Monday till Friday at 1 AM, we would
need the following cronjob:
0 1 * * 1-5 /bin/execute/this/script.sh
 example3
Execute 10 past after every hour on the 1st of every month
10 * 1 * * /bin/execute/this/script.sh
 example4
What if you‘d want to run something every 10 minutes? Well you could
do this:
0,10,20,30,40,50 * * * * /bin/execute/this/script.sh
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But crontab allows you to do this as well:
*/10 * * * * /bin/execute/this/script.sh
 example 5
reboot the system every day at 3 am
0 3 * * * /sbin/reboot
 example 6
Using the @reboot cron keyword, this will execute the specified
command once after the machine got booted every time.
@reboot CMD
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