PhoneCALL Version 2.6
Administrative Manual
PhoneCALL
VecSector , LLC
Revision: 1.0
Section 1: Adding an Account 13
Section 2: Adding an Extension . 15
Section 3: Adding a Phone . 17
Section 4: Configuring Voicemail 21
Section 5: Create a Call Group . 22
Section 1: Call Routing Manager 25
Sub-Section 1: Create New Script/Macro . 26
Sub-Section 2: Incoming Calls . 28
Sub-Section 3: Setup Outgoing Call Rules . 29
Run Selected Script Action . 29
Sub-Section 4: Navigation Menus . 30
Location of Music on Hold files . 32
Listing of Files in Directory . 32
Configuring Global Variables . 36
Section 1: Configuring a new Voice Card . 38
Section 2: Configuring Individual Ports . 39
Section 1 Operating System .. 41
Section 3 Call Detail Records (CDR) 43
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.
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.
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:
The Administration section will contain all the
options on configuring the end-user’s communication to the PBX.
The System Features section will contain the higher-level administration that controls the internal configuration of your PBX.
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.
The Hardware Configuration section will contain all the administrative methods of changing or configuring your telephony hardware inside your PBX.
The Logging section allows you to review any errors or general information about your PBX.
At the top right section of the screen, you will see three options:
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.
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.
Logs off the current user and return to the main login screen.
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.
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.
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.
The authentication username for the account. This is the username the client will use in order to login to their account.
The authentication password for the account. This is the password the client will use in order to login to their account.
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.
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.
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.
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.
This is the matching number dialed from the client.
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.
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.
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.
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.
A phone refers to the physical device(s) that communicate with the server.
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
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.
The Phone Label is the name visible on the front screen of the phone.
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
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.
The configuration of the embedded NIC of the phone.
This is used to route any call marked as ‘emergency’ in the dialplan to go through this server.
The network port used to communicate with the emergency proxy.
In the event the primary VOIP server cannot be reached, the backup server can be tried if available and configured.
The network port used to communicate with the backup proxy.
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.
The network port used to communicate with the outbound proxy.
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
The port used by the phone to listen for SIP messages
The starting RTP session port range
The ending RTP session port range
The phone will processed any SIP headers received with the extra NAT parameters added.
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.
The amount of access given to a telnet session
The prompt to display during a telnet session
The telnet session password
The lines displayed are the amount of lines the phone can support.
You assign an account to each line of the phone.
This is the mailbox name used by the phone system in order to store voicemail messages.
Specify which Tenant the voicemail mailbox belongs to
Enter a name or description for the mailbox. This is also where the directory application looks up people’s names
The PIN used to login to your voicemail mailbox.
Enter the email address you wish to receive voicemail notifications or an attached copy of your voicemail
You can supply a text messaging address where you will receive a text message whenever someone leaves you a voicemail
Specify the tenant the call group belongs to
This is the number matched by the calling party within the system dialplan
Currently, you have to hand enter the dialplan string of which extensions to call. There are no spaces, and each extension is joined with ‘&’
The name of the Time Segment. This is the name that is saved in the phone system dialplan and used for identification.
Used to help describe the purpose of the time segment.
Specify the tenant the call group belongs to
Starting time of the time segment. Time should be in 24hour format (military time)
Ending time of the time segment. Time should be in 24hour format (military time)
Day of the week the time segment starts from
Day of the week the time segment ends
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
This is the system identifier of the script. This will need to be unique for each script.
The name of the script.
Specify the tenant the script belongs to
Just a description of what the script does and some general uses for it.
These are the commands that are matched and ran on the server. This is the core of the script.
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.
The physical port in the phone system.
During what times should this action apply.
The action performed when the line & time segments both apply.
The data needed to perform the action entered.
Description of the Outgoing Call Rule
Specify the Tenant this outgoing call rule applies
The number dialed on the phone, that matches the system dialplan.
The ‘X’ means any number.
Same principle as the extension in Chapter 2.
When
this
dialplan entry is matched, what action should id
perform.
Sub-Section 4
:
Navigation Menus
The name used by the system. Must be unique for each menu.
Specify the tenant the menu belongs to.
Enter the prompt the system will play when the menu is started.
How long to wait after the user stops entering digits.
How long to wait for the user to start entering digits.
Tells the purpose of the menu, and how it should be used.
Actions match the digit(s) pressed from the caller. If a digit combination matches, it will run the performed action.
The name of the action
Specify the name of the tenant the action belongs to
An explanation of what the action does
Which devices will be displayed in the drop-down if
this action is
selected.
Configuring Music on Hold
This will enable the Music on Hold module. If this is unchecked, then all other options will be ignored.
This is the directory where the MP3 files are located.
The default location is ‘/var/lib/asterisk/mohmp3’
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.
Just shows a list of current files in the directory.
Configuring Multi-Site Setup
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.
This is a description of the entry. Just for reference.
The type of connection. This lists the ability of the connection. The default type is ‘friend’.
Specify a specific IP Address for this connection. If an IP is specified, only this IP can use this account.
The password for this account.
Specify a specific context this host can access only
Allow the use of trunking
Creating Audio Files
Specify the directory, filename, type and then click the BROWSE button to upload this audio file to the server
Configuring Remote System Managers
If this option is not checked, all other options are ignored.
Specify the port the manager interface will listen to
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.
Allows you to request which clients are connected to the interface
The name of the Global Variable to be used in scripts. It’s called by ${NAME}
This is the data that will be replaced by the variable.
For adding a description of the variable
The Value is more of a reference for the line itself
Specify the signaling type of the line
….COMING SOON…
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.
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.
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.