IT46 en VoIP4D Laboratory Tutorial AsteriskNOW

7/29/2019 IT46 en VoIP4D Laboratory Tutorial AsteriskNOW

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Practical Guide

How to setup VoIP

Infrastructure using

AsteriskNOW


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Table of Contents

1. Background...........................................................................................................................1

2. The VoIP scenarios...............................................................................................................2

3. Before getting started...........................................................................................................3

3.1 Training Kits...................................................................................................................33.2 Software requirements.....................................................................................................3

3.3 Conventions....................................................................................................................4

3.4 Known issues...................................................................................................................4

4. Virtualization versus dedicated hardware..............................................................................5

5. Installing AsteriskNOW........................................................................................................5

5.1 Installation Screenshots discussed...................................................................................7

6. Configuring AsteriskNOW for Scenario 1 - 2 - 3.................................................................10

6.1 Configuration though the Asterisk GUI Setup Wizard ................................................11

6.1.1 Step 1: Hardware detection...................................................................................11

6.1.2 Step 2: Local extensions settings...........................................................................136.1.3 Step 3: Configuring service providers....................................................................14

6.1.4 Step 4: Outbound calling rules..............................................................................17

6.1.5 Step 5: Voicemail settings.....................................................................................19

6.1.6 Step 6: User extensions.........................................................................................20

6.1.7 Step 7: Incoming calls rules...................................................................................23

6.1.8 Advanced options: Asterisk GUI...........................................................................24

7. Configuration of ATAs........................................................................................................24

8. Quick Installation Guide.....................................................................................................26

8.1 Scenario 1......................................................................................................................26

8.2 Scenario 2......................................................................................................................278.3 Scenario 3......................................................................................................................28

9. Verify your results...............................................................................................................29

9.1 Scenario 1......................................................................................................................29

9.2 Scenario 2......................................................................................................................29

9.3 Scenario 3......................................................................................................................29

1. Background

The first edition of the VoIP-4D Primer, Building voice infrastructure in developing regionsreleased in December 2006 covered the basic aspects of IP Telephony and provided

configuration guidelines for the Asterisk PBX for three basic scenarios. This document aims

to make the installation of such scenarios even easier. While in the first version of the Guide

we configured Asterisk by editing the configuration files, in this guide we are going to use a

graphical user interface (GUI). We have reviewed several initiatives that provide a graphical

interface to Asterisk and decided to prepare this practical tutorial based on the AsteriskGUI

available in Asterisk 1.4.x series.

A new distribution known as AsteriskNOW, includes a straightforward installer and all the

software packages for Asterisk production and development. Although the distribution is still

in beta stage (beta5 in November 2007), it has been designed with a very clean interface anda very intuitive wizard.

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An analysis of other initiatives is available in the document: Making IP telephony knowledge

accessible (prestudy1).

2. The VoIP scenarios

The three scenarios described in this document are similar to the ones described in detailed in

the VoIP-4D Primer.

Scenario 1

Creating a local private telephony network

in a rural community

This scenario consists of a single PBX with

a set of clients. Clients can be either

softphones, VoIP phones or ATAs.

Scenario 2

Interconnecting communities

In this scenario we interconnect two PBXs.

Local extensions of one PBX are made

available to the other and vice versa.

Scenario 3

Connecting communities to the PSTN

In this final scenario, we have

interconnected two PBXs and allow the

possibility of reaching the PSTN from any

of them.

1 Can be downloaded from www.voip4d.org, under Documentation

2
http://www.voip4d.org/http://www.voip4d.org/


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3. Before getting started

If you have not read the VoIP-4D Primer(www.voip4d.org) have a look to the first chapter

as it will provide you with the necessary background to understand the basic concepts of IP

Telephony.

The very minimum hardware requirements for Scenario 1 and 2 is a single PC running

Windows and hosting two virtualized installations of Asterisk. You can test the calls using asoftphone and the voicemail service.

For Scenario 3, you will need two computers, one of them with a dedicated communicationcard TDM400p. Alternatively, you can use two Asterisk appliances such as the IP04s2.

3.1 Training Kits

If you want to run a VoIP training session based on this material, consider at least having

one training kit per group as follows:

2 PCs with network cards 2-4 ATAs or (2-4 VoIP Phones)

2-4 analogue phones (if using ATAs)

1 TDM400p card with 1 FXS and 1 FXO ports

1 4-port switch (better a hub, if you can find one!)

Access to a PSTN line

Alternatively you can use the following training kit 2 IP04 (3 FXS, 1 FX0)

2-4 analogue phones

1 4-port switch

Access to a PSTN line

3.2 Software requirements

ISO Image of AsteriskNOW

VMware Image of Asterisk NOW

http://www.asterisknow.org/downloads

VMware Player

http://www.vmware.com/products/player/

Softphones ; X-Lite, Kiax, etc

http://www.voip-info.org/wiki-Asterisk+IAX+clients

Wireshark (for debugging, advance users)

DHCP Server

2 http://www.rowetel.com/ucasterisk/

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Latest Firefox version >= 2.0.0.9

Documentation

AsteriskNOW quickstart guide

VoIP-4D Primer

This document (Setting up VoIP Infrastructure using AsteriskNOW)

3.3 Conventions

This documentation has used the IP address 192.168.46.135 for the majority of the

screenshots. Be aware that you need to use your own IP addresses for your setup.

1. We will create four local extensions in each PBX, with the names 1000, 2000, 3000,

4000.

2. We will use the same number as username, callerid and password, i.e. username =

callerid = password = 1000 (or 2000, 3000, 4000) ).

3. The voicemail extension is 8500.

4. Scenario 2 and 3 include two different PBXs that should have different IP addresses.

5. Each of the PBXs sees the other PBX as a VoIP Service Provider.

6. The account username: 4646 password: 4646 is created in each of the PBXs for thepurpose of routing calls between them

3.4 Known issues

These are some of the issues found during the preparation of this tutorial:1. If you have problems during authentication, consider using the latest Firefox version

and/or removing the cache and the cookies of your browser.

2. VMplayer can not boot your image if you have a Windows machine with FAT16

filesystem with a size bigger than 2 GB.

Include the line diskLib.sparseMaxFileSizeCheck= "false"at the end of the VMX fileto overcome the problem.

3. AsteriskNOW is still in beta stage. In some cases it is not possible to edit entries after

running the wizard. Consider deleting and recreating the entry instead of editing it.

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4. Virtualization versus dedicated hardware

There are several ways to install AsteriskNOW on a

computer. The method to use depends on your

answers to the following two questions:

1. Does you VoIP setup need to be connected

to the PSTN (TDM support)?

2. Do you have a dedicated machine for the

VoIP setup?

If you answer is Yes to the first question, you need

to install AsteriskNOW in a dedicated machine.

If you do not need to be connected to the PSTN,

you have two options depending on if you have a

machine available for the implementation

(Dedicated machine).

If you do not have a dedicated machine you need to

install VMware player in your machine and the

boot the VMware AsteriskNOW ISO. Thereafter

you can install AsteriskNOW virtually, using your

VMware installation.

If you have a dedicated machine, boot from a CD

that contains the AsteriskNOW ISO.

5. Installing AsteriskNOW Install AsteriskNOW

The distribution is available in three main flavours:

1. A version that runs on the x86, 32-bit/64 bit processors such as Intel P4 and AMD

Athlon XP.

2. A version that runs on the Xen virtual machine.

3. A version that runs on the VMware Player.

If you do not have a dedicated machine available or you want to test the software

distribution, you should consider using the VMware ISO image. Please note that using

the VMware image will not allow you to use any specialized hardware as the PCI

TDM400p card.

Although, it is not mandatory, consider having a DHCP server available on the

network.

Log into the web interface

Open a browser and go to:

https://192.168.46.135

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The web interface provides you access to three main configuration areas:

1. Asterisk GUI configuration wizard

A step-by-step configuration wizard that guide you through 7 steps to get your PBX

up and running

https://192.168.46.135/static/config/setup/install.html

2. The AsteriskGUI

Once you have run the wizard for the first time you can edit and modify the

parameters using the URL

https://192.168.46.135

https://192.168.46.135/static/config/cfgbasic.html

3. The Appliance Platform Configuration Wizard

This wizard allows to configure parameters that are not Asterisk specific, for example

the root password of the system, the IP address, backup schedule, etc.

https://192.168.46.135 :8003/rAA/

Important notice! This tutorial covers only how to use the Asterisk GUI configuration

wizard to set up the scenarios presented. For a complete description of all options available

in the other graphical interfaces, consult the Asterisk Quickstart Guide3.

3 http://www.asterisknow.com/files/downloads/quickstart_asterisknow.pdf

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https://192.168.46.135/static/config/setup/install.htmlhttps://192.168.46.135/https://192.168.46.135/static/config/cfgbasic.htmlhttps://192.168.46.135:8003/rAA/https://192.168.46.135/static/config/setup/install.htmlhttps://192.168.46.135/https://192.168.46.135/static/config/cfgbasic.htmlhttps://192.168.46.135:8003/rAA/


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5.1 Installation Screenshots discussed

Areas with a grey background are advanced tips. If you are not familiar with Asterisk, simplyignore them.

GETTING STARTED - Installing a VMware image

DESCRIPTION

If you install AsteriskNOW using a

ISO image:

Install VMware player and makesure that your Ethernet is in

bridge mode.

By putting the interface in Bridge

Mode, your AsteriskNOW will fetchand IP address via DHCP after

booting.

Important: You need to have a

DHCP server available in your

network.

asterisk-0.9.6.5-x86.vmx

If during the process of booting theVMware player complains about the

size of your FAT filesystem (> 2

GB):

Locate the configuration file of the

guest application, a file that finishes

with VMX.

Use a text editor like Wordpad, and

append an extra line.

CONFIGURATION FILES

Configuration file starts by#!/usr/bin/vmplayer

Append this line:

diskLib.sparseMaxFileSizeCheck= "false"

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GETTING STARTED Make sure AsteriskNOW has an IP address

DESCRIPTION

If your DHCP is working, your

AsteriskNOW will inform you of the

IP address that your box hasobtained.

The AsteriskNOW Console allows

you to update, restart and

shutdown the system.

Using the Console you can also

operate a command line interface

(the asterisk CLI>).

Accessing the box via SSH

You can access the AsteriskNOW

box at any time via SSH.

The account is admin with the

default password password.

You can get admin privileges using

sudo.

CONFIGURATION FILES

If you log into the box using SSH, please have a look

at the /etc/password and /etc/sudores files.You

can see that the user admin can get administrativeprivileges. Consider changing the default password of

the user admin.

It is important to notice that there are 3 different

admin users in each installation:

(1) The admin user that let you log into the box via

SSH.

(2) The admin user that have access to the

AsteriskGUI via HTTPS and

(3) The admin user that can configure the appliance

settings (rPath).

Yes, three different passwords!

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GETTING STARTED Log into AsteriskNOW web interface

DESCRIPTION

Open a browser and go to the IP

address that AsteriskNOW has

obtained.

In our example

http://192.168.46.134

Log into the interface using the user

admin and the default password

password.

/etc/asterisk/manager.conf

AsteriskGUI uses Asterisk manager

commands (Asterisk Manager API)

to communicate with Asterisk.

The user and password of the

AsteriskGUI is available in the

manager.confconfiguration file

CONFIGURATION FILES

[general]displaysystemname = yesenabled = yes

webenabled = yesport = 5038bindaddr = 0.0.0.0[admin]secret = password

read = system,call,log,verbose,command,agent,user,configwrite = system,call,log,verbose,command,

agent,user,config

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6. Configuring AsteriskNOW for Scenario 1 - 2 - 3

In a nutshell, the process of configuring each of the PBXs can be summarized in 7 basic steps:

(Step 1) Verify if any zaptel cardhas been detected.

Any zaptel compatible card should be detected. Cards supported by zaptel include:

Digium, Sangoma, Xorcom Astribank (in beta 6.5), Rhino and Openvox cards.

This step will report no hardware detected if you are configuring a PBX without

zapata compatible hardware or running the VMware version of AsteriskNOW.

Only Scenario 3 will use and detect a zaptel card.

(Step 2) Indicate the first extension number and the length of the localextensions.Here you indicate the number of digits that your local extensions have and what the

first extension number is. In our scenario we will use 4 digits and extension 1000 asthe first one.

(Step 3) Create Service ProvidersIn this step we specify who the service providers of outgoing calls are.

Scenario 1: the PBX is standalone and has no external service providers.

Scenario 2: each PBX sees the other PBX as VoIP service provider.

Scenario 3: the PBX with a TDM card needs to be configured with two different

service providers. The first provider is the other PBX (VoIP) and the second

provider is the Analogue Port of the TDM Card.

(Step 4) Configure (Outbound) Calling Rules

In this step we specify what the calling rules are to reach the different serviceproviders.

Scenario 1: does not need any outbound calling rules.

Scenario 2: need to indicate that to reach the other PBX's local extensions we need

to dial 9 + .

Scenario 3: Same calling rule as in Scenario 2 to reach the other PBX. Also, we

need to add an outgoing calling rule that indicates how to reach the PSTN. To

reach the PSTN, we need to append a 0 to the PSTN number ( 0 + ).

(Step 5) Voicemail extension

In this step we will configure the extension number used to reach the voicemail. Thedefault number for all three scenarios is 8500.

(Step 6) Users extensionsHere we configure all local extensions associated to each of the PBXs. We need to

create four local extensions in all three scenarios (1000, 2000, 3000 and 4000).

The local extensions can be either VoIP clients running IAX or SIP, or analogue ports

if available.

For scenario 2 and 3, we will add the special extension 4646 that is used to route calls

between the PBXs.

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(Step 7) Configure (Inbound) Calling RulesIn Scenario 1 and 2, we do not need to create any special rules for incoming calls as all

calls will be generated locally.

In Scenario 3, we will need to indicate which local extension will ring when there is an

incoming call from the PSTN.

6.1 Configuration though the Asterisk GUI Setup Wizard

This section guides you though the graphical configuration setup wizard provided by

AsteriskNOW. If it is the first time that you log into the AsteriskGUI, you will be redirected

straight to the setup wizard. The setup wizard will guide you through seven (7) steps to

configure your VoIP setup.

This guide includes both basic and advanced configuration tips. Areas with grey background

are advanced tips. If you are not familiar with Asterisk, please ignore them.

6.1.1 Step 1: Hardware detection

STEP 1 HARDWARE DETECTION

(Scenario 1 and 2)

DESCRIPTION

This screenshoot shows Step 1 of

the wizard for Scenario 1 and 2,

where our PBX does not include

any PCI expansion cards.

It is possible to run the wizardagain by accessing the following

URL:

http:///static/config/setup/install.html

CONFIGURATION FILES

All the static web pages of the wizard are available in

the following path:

/var/lib/asterisk/static-http

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STEP 1 HARDWARE DETECTION (2/2)(Scenario 3)

DESCRIPTION

If you have a PCI card as the

TDM400, the wizard will detect the

modules automatically.

In the example, we have 1 FXO and

1 FXS port with the following

functionality:

FXO port: we can attach an external

PSTN line.

FXS port: we can attach a phone.

/sbin/zapscan

The zapscan utility detects the ports

and generates the /etc/zaptel.conf

configuration file.

The configuration files shows the

type of signalling needed for each of

the ports.

fxsks=1 means that port #1 is aFXO that needs FXS Kewlstart

signalling.

CONFIGURATION FILES

#grep -v "#" /etc/zaptel.conf

loadzone = usdefaultzone=usfxsks=1fxoks=2

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6.1.2 Step 2: Local extensions settings

STEP 2 LOCAL EXTENSIONS SETTINGS(Scenario 1, 2 and 3)

DESCRIPTION

In the second step of the

configuration we indicate the lengthof the local extensions. In our setupwe are going to use four digits and

the extension number 1000 as the

first extension.

This configuration is common to all

three Scenarios.

/etc/asterisk/users.conf

This parameter that we set up in

the wizard can be found in the

users.conf with the name userbaseinside of the section [general]

CONFIGURATION FILES

[general]

userbase = 1000localextenlength = 4

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6.1.3 Step 3: Configuring service providers

STEP 3 CONFIGURING SERVICE PROVIDERS (1/3)(Scenario 2 and 3)

DESCRIPTION

In Scenario 2 and 3, we want our

PBX to be able to communicate

with another PBX and vice versa.

To do that, we need to create a new

Service Provider.In this example, we add a new

service provider that is reachable at

the IP address 192.168.46.136.

We indicate that we want tocommunicate using the protocol

IAX using an account with

username 4646and password 4646.This account will be used between

the PBXs for authentication and

routing calls.


/etc/asterisk/extensions.con f

The creation of new service provider

involves:

(1) A new section in the users.conf

file and

(2)A new entry point in the

extensions.conf (dialplan)

In our example we are creating a

service provider [trunk_1] reachable

at 192.168.46.136.

We are using the account user: 4646

password: 4646.

Incoming calls from this provider

fall in the section [DID_trunk_1] of

the dialplan

CONFIGURATION FILES


[trunk_1]disallow =allow = allcallerid =contact =context = DID_trunk_1dialformat = ${EXTEN:1}fromdomain =fromuser =group =hasexten = nohasiax = yeshassip = nohost = 192.168.46.136insecure =port = 4569provider =registeriax = yesregistersip = nosecret = 4646trunkname = Custom - InterIAX Callstrunkstyle = customvoipusername = 4646

/etc/asterisk/extensions.conf[DID_trunk_1]include = default

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STEP 3 CONFIGURING SERVICE PROVIDERS (2/3)(Scenario 3)

DESCRIPTION

If your PBX contains a TDM card

with a FXO port (Scenario 3) we

can communicate with thetelephone network (PSTN).

In the third scenario, you need to

create a new Service Provider

associated to the Analog Port.

In our example, the TDM card

contains a FXO port in slot #1.


/etc/asterisk/extensions.conf

The creation of new service provider

via the PSTN also modifies two

files:

(1) a new section in the users.conffile and

(2) a new entry point in the

dialplan.

In our example the AsteriskGUI

creates a new service provider with

the name [trunk_2] reachable viathe analog port #1

Incoming calls from this provider

fall in the section [DID_trunk_2] of

the dialplan

CONFIGURATION FILES

/etc/asterisk/users.conf[trunk_2]disallow =allow =callerid = asreceivedcontact =context = DID_trunk_2dialformat =fromdomain =fromuser =group = 1

hasexten = nohasiax = nohassip = nohost = dynamicinsecure =port =provider =registeriax =registersip =secret =trunkname = Port 1trunkstyle = analogusername =

zapchan = 1

/etc/asterisk/extensions.conf[DID_trunk_2]include = default

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STEP 3 CONFIGURING SERVICE PROVIDERS (3/3)(Scenario 3)

DESCRIPTION

You can create as many service

providers as you wish.

One of the PBX of the Scenario 3,

has two different Service Providers.

One provider is the other PBX that

can be reached via a VoIP

connection (Custom VoIP) and the

second provider is reachable via the

Analog (TDM400) expansion card.


The configuration file users.confwas introduced in the Asterisk 1.4

series.

In the 1.2.x series, each user or peer

was defined in sip.conf or iax.conf.The entity was classified depending

on the protocol. The users.conf

merges iax.conf, sip.conf and someof the options of zapata.conf intoone single file.

CONFIGURATION FILES

The users.confcontains three types of sections

[general]

This section includes default values.

[trunk_#]

These sections include the configuration of the

different service providers.

[XXXX]These sections include the configuration of the local

extensions (1000, 2000, 3000, 4000). They can be

analog ports or IAX or SIP users.

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6.1.4 Step 4: Outbound calling rules

STEP 4 OUTBOUND CALLING RULES (1/2)(Scenario 2 and 3)

DESCRIPTION

Once we have configured the service

providers we can configure our

dialplan.

In Scenario 2 and 3, we need to create a

rule to be able route calls between the

PBXs. In the example, we create a

calling rule with the name InterIAX

Calls. In this menu, we describe thedialing rules that need to be applied

when we want to reach the extensions of

the VoIP provider (the other PBX) that

we peer with.

We indicate that to reach the other

PBX, we need to dial 9 before the

extension number. To reach the

extension 1000 in the other PBX, we

need to dial 9+1000.


AsteriskNOW allows you to create

different dialplans. The default DialPlan

associated to the context of local

extensions is

numberplan-custom-1.

Outgoing calls between the PBX are

routed using the trundial Macro, that

places a call using:

Dial(IAX2/4646:[email protected]/${EXTEN:1})

and uses the account 4646 data for

authentication.

CONFIGURATION FILES


[numberplan-custom-1]plancomment = DialPlan1include = defaultexten=_9XXXX.,1,Macro(trunkdial,${trunk_1}/${EXTEN:1})comment =_9XXXX.,1,InterIAX Calls,standard

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STEP 4 OUTBOUND CALLING RULES (2/2)(Scenario 3)

DESCRIPTION

In Scenario 3, we need to create

another calling rule that indicatesthat any local extensions can reach

the PSTN by Port #1 (the analog

service provider).

Select Define a custom patternandfill in your outbound calling rule

according to the screenshot to the

right.

In the example, a call is placed by

appending a 0 to a valid PSTN

number, which is defined to be 6or moredigits.


In this example we have two service

providers. The first service provider

is a VoIP provider (another PBX)

and the second provider is the

analog PSTN line.

To reach the VoIP provider: 9 +

extension #

To reach the PSTN via analogue

port: 0 + PSTN #

CONFIGURATION FILES

[numberplan-custom-1]plancomment = DialPlan1include = default;Calls between PBXs. 9 + exten =

_9XXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1})comment = _9XXXX!,1,InterIAX Calls,standard

;Calls to the PSTN. 0 + exten =

_0XXXXXX.,1,Macro(trunkdial,${trunk_2}/${EXTEN:1})comment = _0XXXXXX.,1,outgoing PSTN,standard

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6.1.5 Step 5: Voicemail settings

STEP 5 VOICEMAIL SETTINGS(Scenario 1, 2 and 3)

DESCRIPTION

The default extension for voicemail

is 8500.

The default password for voicemail

is the password of the extension it is

associated with.

When the configuration wizard is

completed, you can change the

password of your voicemail to any

sequence of digits. You will find theoption VW passwordunder Usersin

the main menu.

/etc/asterisk/voicemail.conf/etc/asterisk/users.conf

When voicemail is activated in a

local extension the setting

hasvoicemailis set to yes.

By settings the voicemail we also

modified the way that extensions

are called. Instead of a normal

Dial(), Asterisk 1.4.x will call macro

the [macro-stdexten].

If not other value is specified the

default the Voicemail password is

the same that your account secret.

The vmsecret option allows you to

set a different password for your

voicemail.

In the example extension 3000 uses

the secret 3000 for authentication of

calls and the password 1234 to

reach the mailbox.

CONFIGURATION FILES

[3000]callwaiting = yescid_number = 3000context = numberplan-custom-1email =

fullname = 3000group =hasagent = yeshasdirectory = nohasiax = yeshasmanager = nohassip = yeshasvoicemail = yeshost = dynamicmailbox = 3000secret = 3000threewaycalling = yesvmsecret = 1234

zapchan =registeriax = yesregistersip = yescanreinvite = nonat = nodtmfmode = rfc2833disallow =allow =

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6.1.6 Step 6: User extensions

STEP 6 USER EXTENSIONS (1/3)(Scenario 1, 2 and 3)

DESCRIPTION

In this step we will create the four local

user extensions. The extensions can be

associated to an IAX or SIP device such

as an ATA or VoIP Phone, or associated

to a analogue port available in the PBX.


Each of the new extensions will have

entry of the type [1000], [2000], [3000],

etc.

If the local extension is a SIP or IAX

device it will be indicated with the

values:

hassip = yeshasiax = yes

CONFIGURATION FILES

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STEP 6 USER EXTENSIONS (2/3)(Scenario 2 and 3)

DESCRIPTION

In Scenario 2 and 3 we need to create

extension 4646, which needs to beavailable in both PBXs.

This extension is created to accept

incoming calls from the other PBX.


This extension is not visible to the

users and it is used for the purpose of

routing and authenticating calls between

the PBXs.

In Scenario 3 we are using IAX as the

protocol for interconnecting the PBXs.

IAX is more NAT friendly and efficient

in terms of bandwidth.

CONFIGURATION FILES

[4646]callwaiting = yescid_number = 4646context = numberplan-custom-1email =fullname = 4646group =hasagent = yeshasdirectory = nohasiax = yeshasmanager = nohassip = yeshasvoicemail = yeshost = dynamicmailbox = 4646secret = 4646threewaycalling = yesvmsecret =zapchan =registeriax = yesregistersip = yescanreinvite = nonat = no

dtmfmode = rfc2833disallow =allow =

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STEP 6 USER EXTENSIONS (3/3)

(Scenario 3)

DESCRIPTION

In Scenario 3, we need to define

which local extension that should beassociated with the Analog Port #2

(the phone). In this example, we

have chosen Extension 1000 for that

task.

We define the association by editing

the existing User Extension for

Extension 1000 and select Analog

Port #2 as Analog Phone.


Although it might look surprising, it

is possible to have an extension

associated to more than one

communication technology.

In the example, extension 1000 is

reachable in the Analogue Port #2

andvia SIP and IAX.

hasiax = yeshassip = yeszapchan = 2

This allows us to have as many as

three devices associated to the same

extension number. The three

devices will ring simultaneously.

CONFIGURATION FILES

[1000]callwaiting = yescid_number = 1000context = numberplan-custom-1email =fullname = 1000group =hasagent = nohasdirectory = no

hasiax = yeshasmanager = nohassip = yeshasvoicemail = yeshost = dynamicmailbox = 1000secret = 1000threewaycalling = yeszapchan = 2registeriax = yesregistersip = yescanreinvite = nonat = no

dtmfmode = rfc2833

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6.1.7 Step 7: Incoming calls rules

STEP 7 - INCOMING CALLS RULES(Scenario 3)

DESCRIPTION

In Scenario 3, we need to decide

what to do with the calls originated

in the PSTN.

In the example, we indicate that all

calls from the PSTN should be

forward to the local extension 1000.



Port #1 is a FXO port connected to

the PSTN (zapchan = 1).

Incoming calls fall in the context

DID_trunk_2.

In the dialplan, under the context

[DID_trunk_2] we see that by

default all calls (_X.,s) are forward

to extension 1000

Goto(default|1000|1)

CONFIGURATION FILES

[trunk_2]disallow =allow =callerid = asreceivedcontact =context = DID_trunk_2

dialformat =fromdomain =fromuser =group = 1hasexten = nohasiax = nohassip = nohost = dynamicinsecure =port =provider =registeriax =registersip =

secret =trunkname = Port 1trunkstyle = analogusername =zapchan = 1

[DID_trunk_2]include = defaultexten = _X.,1,Goto(default|1000|1)exten = s,1,Goto(default|1000|1)

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6.1.8 Advanced options: Asterisk GUI

ADVANCED OPTIONS ASTERISK GUI

DESCRIPTION

After completing the seven steps you

can have a look to the AsteriskGUI

interface.

This interface allows to modify your

entries via the wizard and create

more advance services.

When you have made changes to the

configuration, do not forget to press

the button Active Changes, in order

apply the changes.

7. Configuration of ATAs

No matter which ATA or IP Phone you need to configure, you will find that they can beconfigured in a similar manner. This example shows the configuration process of a Linksys

PAP2 Internet Phone Adapter. The configuration is the same for a Sipura (SPA-3000).

IP settingsThe ATA needs to have an IP address in order to be able to communicate with other devices

on the LAN or the Internet. The IP address can be static or dynamic. In this example, we

have chosen to obtain an IP address through DHCP.

All IP settings of the ATA are configured using the handset.

1. Attach an analog phone to the ATA

2. Connect the ATA to the LAN where the DHCP is running

3. Enter the configuration menu of the ATA by pressing **** on the phone.

4. Enable DHCP by pressing 101# followed by 1.

5. Make sure that the ATA has obtained an IP address by pressing 110#.

Extension numberThe extension number of the ATA is configured through its web interface. Direct your browser

to http:///admin/advanced

Go to the tab Line 1, and fill in the following fields:

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Proxy and RegistrationProxy: 192.168.46.135 (the IP address of the PBX you want to register)

Register: Yes

Subscriber InformationDisplay Name: 1000

User ID: 1000

Password: 1000Use Auth ID:yes

Auth ID: 1000

You can verify from the web based Asterisk Configuration Panel that the ATA is registered in

the PBX.

1. Go to Asterisk CLI in the left menu

2. On the bottom of the page (in the pink text field), writesip show peers

3. All registered phones and ATAs will be listed with IP address and extension number.

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8. Quick Installation Guide

This section includes a 7-step quick installation guide for Scenario 1, 2 and 3. Please note that

the red crosses in the table indicate steps in the configuration procedure that are not needed

for that specific scenario.

8.1 Scenario 1

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8.2 Scenario 2

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8.3 Scenario 3

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9. Verify your results

This section includes a set of Checkpoints for each Scenario that you should be able to do

with your current VoIP setup. If you successfully manage all checkpoints listed for your

Scenario, your Asterisk based VoIP setup has been configured correctly.

9.1 Scenario 1

Checkpoint 1: Place local phone calls from one extension to another within the same PBX.

Checkpoint 2: Call a local extension and leave a voice messages (don't pick up the phone!).Use the voicemail to fetch the voice message you just left.

9.2 Scenario 2

Checkpoint 1: Place local phone calls from one extension to another within the same PBX.

Checkpoint 2: Place phone calls between the two PBX's by using the prefix 9 before theextension number.


9.3 Scenario 3

Checkpoint 1: Place local phone calls from one extension to another within the same PBX.Checkpoint 2: Place phone calls between the two PBX's by using the prefix 9 before theextension number.


Checkpoint 4: Call to the PSTN from any of the PBXs (try both).

Checkpoint 5: Call in to the PBX from the PSTN.

Documents

IT46 en VoIP4D Laboratory Tutorial AsteriskNOW