PhoneCALL Administrative Manual



Table of Contents



1 Introduction to PhoneCALL 2 Chapter 1 3 Chapter 2 4 Chapter 3 5 Chapter 4 6 Chapter 5

PhoneCALL Version 2.6

Administrative Manual

 

 

 

PhoneCALL

 

 

 

VecSector , LLC

Revision: 1.0


Introduction to PhoneCALL . 6

Overview .. 6

Terminology Used . 6

Chapter 1 . 8

The PhoneCALL interface . 8

Administration . 8

System Features . 9

Tenant Management 9

Hardware Configuration . 10

Hardware Configuration . 10

Logging . 10

System Preferences . 10

Preferences . 10

Save Config . 11

Logoff 11

Chapter 2 . 12

General Administration . 12

Section 1:    Adding an Account 13

NAME . 13

DESCRIPTION .. 13

TENANT . 13

USERNAME . 13

PASSWORD .. 13

CALLERID .. 14

SCRIPT . 14

TECHNOLOGY .. 14

ADMIN NOTE: 14

Section 2: Adding an Extension . 15

Extension . 15

Tenant 15

Description . 15

Script 15

Arguments . 16

Section 3: Adding a Phone . 17

Phone Type . 17

MAC Address . 18

Phone Label 18

DialPlan . 18

TimeLine . 18

Network Media Type . 18

Emergency Proxy Address . 18

Emergency Proxy Port 18

Backup Proxy Address . 18

Backup Proxy Port 18

Outbound Proxy Address . 19

Outbound Proxy Port 19

Enable NAT . 19

VOIP Control Port 19

Start Media Port 19

End Media Port 19

NAT Received Processing . 19

VAD .. 19

Telnet Level 19

Phone Prompt 20

Phone Password . 20

Lines . 20

Section 4: Configuring Voicemail 21

Mailbox . 21

Tenant 21

Username . 21

Password . 21

Email Address . 21

Text Message Address . 21

Section 5: Create a Call Group . 22

Tenant 22

Call Group Extension . 22

Enter Numbers . 22

Section 6: Time Segments . 23

Name . 23

Description . 23

Tenant 23

Start Time . 23

End Time . 23

Start Day . 23

End Day . 23

Chapter 3 . 24

Advanced Administration . 24

Section 1: Call Routing Manager 25

Sub-Section 1: Create New Script/Macro . 26

Macro Name . 26

Name . 26

Tenant 26

Description . 26

Commands . 27

Arguments . 27

Sub-Section 2: Incoming Calls . 28

Line . 28

Time Segment 28

Action . 28

Data . 28

Sub-Section 3: Setup Outgoing Call Rules . 29

Name Description . 29

Tenant 29

If Call String Matches . 29

Run Selected Script Action . 29

Sub-Section 4: Navigation Menus . 30

Menu Name . 30

Tenant 30

Play Prompt 30

Digit Timeout 30

Wait for Response . 30

Description . 30

Actions . 30

Add Action . 31

Music-On-Hold . 32

Enable Music on Hold . 32

Location of Music on Hold files . 32

Upload File . 32

Listing of Files in Directory . 32

Multi-Site Setup . 33

Name . 33

Description . 33

Type . 33

Host 33

Secret 33

Context 33

Trunk . 33

Creating Audio Files . 34

Upload File . 34

Remote System Manager 35

Enabled . 35

Port 35

Bind Address . 35

Display Connects . 35

Configuring Global Variables . 36

Name . 36

Data . 36

Description . 36

Chapter 4 . 37

Hardware Configuration . 37

Section 1: Configuring a new Voice Card . 38

Line . 38

Signal Type . 38

Section 2: Configuring Individual Ports . 39

Chapter 5 . 40

Logging . 40

Section 1 Operating System .. 41

Section 2 Phone System .. 42

Section 3 Call Detail Records (CDR) 43

 


1 Introduction to PhoneCALL

 

1.1 Overview

 

PhoneCALL is a PHP/SMARTY based web GUI software package designed to configure and administer the Asterisk open-source PBX.   Within PhoneCALL, you will be able to quickly perform all the administrative tasks that are needed from telephony administrators.

 

1.2 Terminology Used

 

PhoneCALL was written to help make telephony administration easy, but some understanding of the terminology used in throughout the process will only make administration easier.   The more you learn about VOIP/Telephony, the easier the process will be.   Below are some very common terms used throughout the administration process with a brief description of each term.   If you still do not have a complete understanding of the terminology, we recommend learning more about each term before proceeding.

 

An extension is the number or name reserved on the PBX for internal call routing.   Thing of an extension as a username.   Each device(phone) connected to the PBX typically has an extension associated with it.

A phone is the communication device that connects you with the PBX and/or the remote party.   A phone can be a hardphone, a tangible piece of hardware, or a softphone – a piece of software written to function like a phone.  

Or Dual Tone Multi-Frequency Tones.   A ‘DTMF’ is the sound generated when a number is pressed on the number pad for calling.

Or ‘public switched telephone network’.    PSTN is referred to as the analog phone lines coming into your PBX from the telephone company.

Or ‘ i ntegrated services digital network’.   ISDN refers to the sending of voice, data, video over digital telephone lines or analog telephone wires.

Type of ISDN that consists of having two(2) 64k B-Channels(the actual line) and one D-Channel(used for signaling).

Type of ISDN that consists of up to 23 (in the US – 30 in Europe )   B -Channels(actual line) and one(1) D-Channel(signaling)

A communication channel between two points.   It is usually referred whenever you speak of a large amount of lines interconnected.

A circuit connection two devices.   ISDN or T1 could both be referred as ‘lines’ along with a POTS (Plain Old Telephone Service) analog line.

Or “Session Initiated Protocol”.   It is a type of signaling protocol supported by many VOIP phones.   SIP is the protocol used to setup the call between your phone & the PBX.

Or “Inner-Asterisk Exchange”.   It can be used as a signaling protocol from the phone to the PBX (like SIP) or from used to create trunks between Asterisk servers.

Zaptel is the type of hardware card in the Asterisk server.   The term was shortened to ‘ZAP’ to refer to the PSTN line connecting to the Zaptel hardware.

List of callers that are left waiting until a caller can answer them.

Contexts are used to organized sections together.   It allows you to group areas of configurations together.   Example:

                [ incoming]

                                Exten => s ,1,Answer()

                [ outgoing]

                                Exten => _911 ,1,Dial(Zap/1/911)

The incoming context is separate from the outgoing context and vice versa.   You can also include one context into another.

A set of commands to perform an action

Or “i nteractive voice response”.   This is referred by a caller using a touch-tone telephone to interact with a menu.   Example:

                ‘Welcome to XYZ Company’

                                Press ‘1’ for Sales

                                Press ‘2’ for technical support

Or “Foreign Exchange Office”.   FXO interfaces are used to connect the PSTN to your PBX

Or “Foreign Exchange Station”.    FXS interfaces are used to connect to devices such as a fax machine to your PBX.

A dialplan is what is used to control where a call goes and what it should do.


2 Chapter 1

 

2.1 The PhoneCALL interface

 

So you are ready to start using the PhoneCALL interface!   The first thing you will need to do is open a web browser and type in the IP Address of your PhoneCALL PBX.   You will see the following screen:

 

 

 

The account defaults loaded with PhoneCALL are:

 

Username:         admin

Password:         admin

 

 

It’s highly recommended to change the ‘admin’ username after logging in.

 

Once logged into PhoneCALL, you will be greeted with a system overview screen along with lots of options on the left-hand side of the screen as seen here:

 

Each section is divided by function:

 

2.1.1 Administration

 

The Administration section will contain all the options on configuring the end-user’s communication to the PBX.

 

2.1.2 System Features

 

The System Features section will contain the higher-level administration that controls the internal configuration of your PBX.

 

 

 

2.1.3 Tenant Management

 

Allows you to separate the different businesses that will utilize the system.   Each Tenant has their own set of extensions and scripts that are separate from the others.   This will allow several business owners use one server to consolidate costs, but allow for their own customized setup as if they had their own server.

 

 

2.1.4

Hardware Configuration

 

The Hardware Configuration section will contain all the administrative methods of changing or configuring your telephony hardware inside your PBX.

 

 

2.1.5 Logging

 

The Logging section allows you to review any errors or general information about your PBX.

 

 

2.1.6 System Preferences

 

At the top right section of the screen, you will see three options:

 

 

2.1.6.1 Preferences

Allows you to configure the System-wide PhoneCALL preferences.

 

 

Site Name

Changes the name of the site.

Row Heading Color

Changes all row headers to the specified color

Disable Security Image

Toggles the login security image for added security.   It is disabled by default, but we recommend it to be enabled for PBX’s that can be accessed from the internet.

Disable Logins (Site Offline)

This option will allow the Administrator to turn-off the PhoneCALL GUI to all non-administrators.   Useful for upgrades.

 

2.1.6.2 Save Config

Clicking this option will write all of the database information into flat-files into the /etc/asterisk and / tftpboot directories.   All files in the /etc/asterisk directory are to be parsed by Asterisk and used for PBX configuration.   All files in / tftpboot are used by tftp-enabled Phones to update their software configuration.

 

2.1.6.3 Logoff

Logs off the current user and return to the main login screen.

 


3 Chapter 2

 

3.1 General Administration

 


3.1.1 Section 1:  
Adding an Account

 

3.1.2  

3.1.2.1 NAME

Name is the header configuration for the account.   This will need to be unique for each entry.   It is the part of the account that is inclosed in [] in the configuration file, or what we refer as – the header.

 

3.1.2.2 DESCRIPTION

Description doesn’t affect the configuration of the extension.   It is there for your reference to make notes about the account and possibly the reason for adding the account.

 

3.1.2.3 TENANT

Choose the Tenant this extension will be granted to.   A single account can only be assigned to a single Tenant within version 2.6.   This may change in future versions.

 

3.1.2.4 USERNAME

The authentication username for the account.   This is the username the client will use in order to login to their account.

 

3.1.2.5 PASSWORD

The authentication password for the account.   This is the password the client will use in order to login to their account.

 

3.1.2.6 CALLERID

This option will let you override the callerid supplied by the client.   If no callerid is supplied, the callerid supplied by the client will be used.

 

3.1.2.7 SCRIPT

This is a quick-link of creating a system dialplan entry for this account.

A dialplan extension is how one phone calls another phone.   Without a system dialplan entry for the account, no one will be able to directly call this extension from another account.

 

3.1.2.8 TECHNOLOGY

The Technology refers to the communication between the phone system and the remote client.   If you have a SIP-based client, then you would need to make this account as a SIP account.

 

 

3.1.2.9 ADMIN NOTE:

After all parameters are filled, pressing submit will add the Account to the phone system.   Performing a LIST of the accounts will show you the account and allow you   to edit the account for errors.   The account is not active until you run ‘SAVE CONFIG’ and reload the system.


3.1.3 Section 2 :
Adding an Extension

 

3.1.4  

3.1.4.1 Extension

This is the matching number dialed from the client.

 

3.1.4.2 Tenant

Assign the extension to a tenant.   As of 2.6 – one extension can only be applied to one tenant at a time.   Another words, we couldn’t assign the extension 2165 in the above figure to multiple tenants.

 

3.1.4.3 Description

Description doesn’t affect the configuration of the extension.   It is there for your reference to make notes about the account and possibly the reason for adding the extension.

 

3.1.4.4 Script

Think of the script as an application that will run whenever this extension is called.   In the example above, the script ‘EXTENSION WITH VOICEMAIL’ was chosen.   This particular script calls for 2 arguments:   an extension to call, and a voicemail box to go to.

 

Scripts give the system a great amount of flexibility.   You can create an unlimited amount of scripts or browse our online repository of scripts that change the behavior of an extension.

 

3.1.4.5 Arguments

Arguments are only shown if a particular script needs further information in order to function properly.   The arguments will appear after submitting the form with the new script assigned.


3.1.5 Section 3 :
Adding a Phone

 

 

A phone refers to the physical device(s) that communicate with the server.  

 

3.1.5.1 Phone Type

Selecting the phone type will change the fields prompted that are relevant to that phone’s configuration.   Currently, 2.6 only has support for Cisco phones.   More phone templates will be added in the next release

 

3.1.5.2 MAC Address

The MAC address is used as the configuration filename of the device.   Since MAC addresses should be unique for every network device on your network, phone vendors chose to use the MAC address as the method of remotely configuring devices on the network.  

 

3.1.5.3 Phone Label

The Phone Label is the name visible on the front screen of the phone.

 

3.1.5.4 DialPlan

This is the dialplan for this particular phone.   A dialplan for a phone is the configuration of the way the digits are handled.   It also provides pattern-matching, so if you add a dialplan configuration to match a 4-digit extension, the user will not have to press ‘DIAL’ in order to complete the call

 

3.1.5.5 TimeLine

This is the TimeZone of the phone.   This is used to determine what time should display on the phone, and if daylight savings time will be used.

 

3.1.5.6 Network Media Type

The configuration of the embedded NIC of the phone.  

 

3.1.5.7 Emergency Proxy Address

This is used to route any call marked as ‘emergency’ in the dialplan to go through this server.

 

3.1.5.8 Emergency Proxy Port

The network port used to communicate with the emergency proxy.

 

3.1.5.9 Backup Proxy Address

In the event the primary VOIP server cannot be reached, the backup server can be tried if available and configured.

 

3.1.5.10 Backup Proxy Port

The network port used to communicate with the backup proxy.

 

3.1.5.11 Outbound Proxy Address

Assists phones that are unable to leave an internal network due to security (requires onsite outbound proxy) or to assist a remote phone behind a NAT’d network find the remote extension/server.   If enabled, all requests are sent to the Outbound Proxy instead of the Primary Proxy.

 

3.1.5.12 Outbound Proxy Port

The network port used to communicate with the outbound proxy.

 

3.1.5.13 Enable NAT

Check this option in order to allow the phone send NAT headers in it’s communication

NAT Wan Address

If the phone is unable to determine it’s external IP address, you can specify the external/public IP manually

 

3.1.5.14 VOIP Control Port

The port used by the phone to listen for SIP messages

 

3.1.5.15 Start Media Port

The starting RTP session port range

 

3.1.5.16 End Media Port

The ending RTP session port range

 

3.1.5.17 NAT Received Processing

The phone will processed any SIP headers received with the extra NAT parameters added.  

 

3.1.5.18 VAD

If no voice is being transmitted, the audio stream is stopped until someone speaks.   This helps with networks with a large amount of lag.   This can results in some choppy audio and cut off of the beginning of a sentence.

 

3.1.5.19 Telnet Level

The amount of access given to a telnet session

 

3.1.5.20 Phone Prompt

The prompt to display during a telnet session

 

3.1.5.21 Phone Password

The telnet session password

 

3.1.5.22 Lines

The lines displayed are the amount of lines the phone can support.

You assign an account to each line of the phone.

3.1.6 Section 4 :
Configuring Voicemail

 

 

3.1.6.1 Mailbox

This is the mailbox name used by the phone system in order to store voicemail messages.

 

3.1.6.2 Tenant

Specify which Tenant the voicemail mailbox belongs to

 

3.1.6.3 Username

Enter a name or description for the mailbox.   This is also where the directory application looks up people’s names

 

3.1.6.4 Password

The PIN used to login to your voicemail mailbox.

 

3.1.6.5 Email Address

Enter the email address you wish to receive voicemail notifications or an attached copy of your voicemail

 

3.1.6.6 Text Message Address

You can supply a text messaging address where you will receive a text message whenever someone leaves you a voicemail

 


 

3.1.7 Section 5 :
Create a Call Group

 

 

 

3.1.7.1 Tenant

Specify the tenant the call group belongs to

 

3.1.7.2 Call Group Extension

This is the number matched by the calling party within the system dialplan

 

3.1.7.3 Enter Numbers

Currently, you have to hand enter the dialplan string of which extensions to call.   There are no spaces, and each extension is joined with ‘&’

 

 


3.1.8 Section 6 :
Time Segments

 

 

 

3.1.8.1 Name

The name of the Time Segment.   This is the name that is saved in the phone system dialplan and used for identification.

3.1.8.2 Description

Used to help describe the purpose of the time segment.

3.1.8.3 Tenant

Specify the tenant the call group belongs to

3.1.8.4 Start Time

Starting time of the time segment.   Time should be in 24hour format (military time)

3.1.8.5 End Time

Ending time of the time segment.   Time should be in 24hour format (military time)

3.1.8.6 Start Day

Day of the week the time segment starts from

3.1.8.7 End Day

Day of the week the time segment ends


 

4 Chapter 3

 

4.1 Advanced Administration

 


4.1.1 Section 1 :
Call Routing Manager

 

 

 

The Call Routing Manager is the engine behind the system extensions ( dialplan).

Whenever someone dials an extension that matches the system dialplan, a script is ran to control what should happen to that call.

 

The Script/Macro editor is where you will create these scripts and their possible required arguments.

 

The Incoming Calls section is how to handle the incoming calls per port.

 

The Outgoing Calls is how the system handles an outbound call to the PSTN.

 

Navigation Menus can range from IVR menus, to automated attendants, and even calling card applications.   Anything that requires a prompt to play and requires input from the user.
Sub-Section 1 :
Create New Script/Macro

 

 

4.1.1.0.1 Macro Name

This is the system identifier of the script.   This will need to be unique for each script.

 

4.1.1.0.2 Name

The name of the script.  

 

4.1.1.0.3 Tenant

Specify the tenant the script belongs to

 

4.1.1.0.4 Description

Just a description of what the script does and some general uses for it.

 

4.1.1.0.5 Commands

These are the commands that are matched and ran on the server.   This is the core of the script.

 

4.1.1.0.6 Arguments

Clicking the ‘ADD ARGUMENT’ button will add an argument for the script.   If you notice in the Commands section, some of the variables have ${ARG1}.   That means it uses the information taken from argument 1.  


4.1.1.1 Sub-Section 2 :
Incoming Calls

4.1.1.2  

4.1.1.2.1 Line

The physical port in the phone system.

4.1.1.2.2 Time Segment

During what times should this action apply.

4.1.1.2.3 Action

The action performed when the line & time segments both apply.

4.1.1.2.4 Data

The data needed to perform the action entered.


4.1.1.3 Sub-Section 3 :
Setup Outgoing Call Rules

4.1.1.4  

4.1.1.5  

4.1.1.6 Name Description

Description of the Outgoing Call Rule

 

4.1.1.6.1 Tenant

Specify the Tenant this outgoing call rule applies

 

4.1.1.6.2 If Call String Matches

The number dialed on the phone, that matches the system dialplan.

The ‘X’ means any number.

 

4.1.1.6.3 Run Selected Script Action

Same principle as the extension in Chapter 2.   When this dialplan entry is matched, what action should id perform.
Sub-Section 4 :
Navigation Menus

 

 

 

4.1.1.6.4 Menu Name

The name used by the system.   Must be unique for each menu.

4.1.1.6.5 Tenant

Specify the tenant the menu belongs to.

4.1.1.6.6 Play Prompt

Enter the prompt the system will play when the menu is started.

4.1.1.6.7 Digit Timeout

How long to wait after the user stops entering digits.

4.1.1.6.8 Wait for Response

How long to wait for the user to start entering digits.

4.1.1.6.9 Description

Tells the purpose of the menu, and how it should be used.

4.1.1.6.10 Actions

Actions match the digit(s) pressed from the caller.   If a digit combination matches, it will run the performed action.

 


4.1.1.6.11 Add Action

 

4.1.1.6.11.1 Action Name

The name of the action

 

4.1.1.6.11.2 Tenant

Specify the name of the tenant the action belongs to

 

4.1.1.6.11.3 Description

An explanation of what the action does

 

4.1.1.6.11.4 Map to Device

Which devices will be displayed in the drop-down if this action is selected.

4.1.2 Music-On-Hold

 

Configuring Music on Hold

 

4.1.2.0.1 Enable Music on Hold

This will enable the Music on Hold module.   If this is unchecked, then all other options will be ignored.

4.1.2.0.2 Location of Music on Hold files

This is the directory where the MP3 files are located.

The default location is ‘/var/lib/asterisk/mohmp3’

4.1.2.0.3 Upload File

Type the name of the file, choose the file type.   Then click the ‘BROWSE’ button in order to choose the file you are uploading to the server in the location listed above.

4.1.2.0.4 Listing of Files in Directory

Just shows a list of current files in the directory.

 


 

4.1.3 Multi-Site Setup

 

Configuring Multi-Site Setup

 

 

4.1.3.0.1 Name

The name field is used as the header id of located within the ‘ iax.conf’ file.   It’s also used as the username.   No spaces for this field would be best.

4.1.3.0.2 Description

This is a description of the entry.   Just for reference.

4.1.3.0.3 Type

The type of connection.   This lists the ability of the connection.   The default type is ‘friend’.

4.1.3.0.4 Host

Specify a specific IP Address for this connection.   If an IP is specified, only this IP can use this account.

4.1.3.0.5 Secret

The password for this account.

4.1.3.0.6 Context

Specify a specific context this host can access only

4.1.3.0.7 Trunk

Allow the use of trunking

4.1.4 Creating Audio Files

 

Creating Audio Files

 

 

4.1.4.0.1 Upload File

Specify the directory, filename, type and then click the BROWSE button to upload this audio file to the server


 

4.1.5 Remote System Manager

 

Configuring Remote System Managers

 

 

4.1.5.0.1 Enabled

If this option is not checked, all other options are ignored.

4.1.5.0.2 Port

Specify the port the manager interface will listen to

4.1.5.0.3 Bind Address

Specify a specify address of the server that you want the manager interface to listen to.   Recommended for sites with multiple NICs that may have a public IP assigned.   Leaving the IP of ‘0.0.0.0’ will allow the manager interface to connect from all interfaces.

4.1.5.0.4 Display Connects

Allows you to request which clients are connected to the interface


 

4.1.6 Configuring Global Variables

 

4.1.6.0.1 Name

The name of the Global Variable to be used in scripts.   It’s called by ${NAME}

4.1.6.0.2 Data

This is the data that will be replaced by the variable.

4.1.6.0.3 Description

For adding a description of the variable


4.1.7  

5 Chapter 4

 

5.1 Hardware Configuration


5.2  

5.2.1 Section 1 :
Configuring a new Voice Card

 

5.2.1.0.1 Line

The Value is more of a reference for the line itself

5.2.1.0.2 Signal Type

Specify the signaling type of the line


 

 

5.2.2 Section 2 :
Configuring Individual Ports

 

….COMING SOON…

 

6 Chapter 5

 

6.1 Logging


 

6.1.1 Section 1
Operating System

 

The logs in this section show various Operating System logs.   These logs are helpful for debugging any non-PBX issues such as services, login attempts to the server.

 

These logs will later include more in-depth debugging information in the event you are having trouble with the PBX, and need to send in a support request.

6.1.2 Section 2
Phone System

 

These logs are related to the PBX itself.

 

These logs will later include more in-depth debugging information in the event you are having trouble with the PBX, and need to send in a support request.

 

6.1.3 Section 3
Call Detail Records (CDR)

 

The CDR interface shows the logs for all of the calls in/out of the system.

We’ll be adding a method of call tracing and call statistics soon.