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EdgeBOX V 4.3 VoIP – How To
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1 VoIP accounts and their settings....................................................................................................................3
1.1Two methods of creating a VoIP account on the edgeBOX....................................................................3
1.3 Advanced settings.................................................................................................................... ..............5
1.4 Codecs.................................................................................................................................... ...............6
1.5 Security.............................................................................................................................. ....................6
2 Regisering Phones on the edgeBOX.............................................................................................................7
2.1 Softphone (x-Lite)............................................................................................................................... ....7
2.2 VoIP Phone ...................................................................................................................... .....................9
3 The LCR and Outgoing calls........................................................................................................................10
3.1 VoIP Providers, SIP Proxies, and Soft Switches..................................................................................13
3.2 Remote Switches ................................................................................................................................15
3.3 Calls to ISDN................................................................................................................................... .....15
3.4 Calls to Primary cards E1/T1...............................................................................................................16
3.5 Analog lines................................................................................................................................ ..........16
3.6 Authentication................................................................................................................................... ....17
4 Incoming calls.......................................................................................................................... ....................18
4.1 The IVR............................................................................................................................... .................18
4.2 Internal ............................................................................................................................ ....................20
4.3Call Rules.................................................................................................................................... ..........21
4.4 DID Routes...................................................................................................................................... .....21
4.5 DID Ranges .................................................................................................................................... .....22
4.6 Sound Manager................................................................................................................................ ....22
5 The PBX................................................................................................................................................ .......22
5.1 Creating agents........................................................................................................................... .........22
5.2 Creating queues ......................................................................................................................... .........23
5.3 Huntgroups...................................................................................................................................... .....24
5.4 Confrences, Parking and Voicemail.....................................................................................................25
6 Resources.......................................................................................................................................... ..........27
6.1 CDR files............................................................................................................................... ...............27
6.1 Sites................................................................................................................................. ....................28
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1. VoIP accounts and their settings
VoIP accounts must be configured on the edgeBOX in order for users to access the VoIP
functionality. There are two kinds of accounts SIP (session Initiation Protocol) and IAX2 (InterAsterisk exchange protocol). Both are standard protocols for voice over IP technology.
1.1 Two methods of creating a VoIP account on the edgeBOX
There are two ways to create a VoIP account one is through regular user creation and the other is
through VoIP services. We will be discussing creating a VoIP account via both methods; however
we recommend the use of one over the other depending on the usage of the edgeBOX. If the VoIP
account will be used specifically for a device and self administration is not needed then you may
create the phone via the VoIP & IPPBX area of the admin GUI. However if the account will be used
by a person then we would recommend creating the account via the NAC menu as this will give the
user access to call forwarding, voicemail, and a few extra features discussed later in this document.
Regular user creation via NAC:
Log into the admin graphic user interface (GUI) and go to NAC choose Users then List. Click on
New.
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1. Username: This is the user id you will give
to your user.
2. First Name: This is the user’s first name.
3. Last Name: This is the user’s last name.
4. Access profile: This is the access profile the
user will belong too.
5. Password and Confirm password: this is
where you enter the user’s password for
other services. (Please see manual for
more information on NAC).
6. Accesses you must check VoIP in order for
the user’s extension to be created.
7. Extension Number: This will be the
extension the user will have.
8. Extension Password: This is the password
used to log into the extension.
9. Authentication Pin: This is a unique number
the user will need to make any call other
than free calls if VoIP authentication
(discussed later) is turned on.
10. Permissions: These are the type of calls
you can give users access too. They can be
any of the following: Local, Long Distance,
Mobile, International, and All types of calls. These permissions are cumulative as if you
give a user access to make mobile calls he/she will have access to local and long distance
also.
Creation of a VoIP only account:
Log into the admin graphic user interface and click New.
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1. Protocol: You may choose what protocol (SIP or
IAX2) you would like to use for your VoIP account.
This choice should be made with consideration of
the phone or soft phone’s requirements.
2. Extension Name: This is simply the unique name
you will assign an extension.
3. Extension Number: This field is the unique
number assigned to the extension, the number
other users would call.
4. Password: This is simply the password for this
extension.
5. Publish Extension: This allows you to dial this
phone directly (via remote switch) if the trunk is set
to untrusted. If the line is trusted, all phones can be
contacted directly via the trunk, whether this option
is checked or not.
6. Active Voicemail: This option allows you to
activate Voicemail for this phone.
7. Voicemail PIN: Here you can select the Voicemail
PIN number.
8. Email Address: In this area you fill in the email
address of the person you would like notified when
a voicemail arrives.
9. Voicemail Timeout: Select how long this phone
will ring before voicemail is activated.
1. Caller ID: This field is where the caller ID
information is placed
2. Nat: You can configure this phone t use Nat
if for example it will be registering from
another network.
3. MD5 Hash: This functionality is supported
with the SIP protocol only and send the
password encrypted in MD5 hash.
4. Host: If static is selected, you will need to
specify the IP address of the phone, otherwise
the phone will provide the IP address when
registering.
5. Default IP: This is where you would place
the IP if static is chosen in Host.
6. Port: The default ports are 5060 for SIP and
4059 for IAX2.
7. DTMF: This deals with the way the phone
deals with signaling, the choices are INBAND
(not supported by the GSM codec), rfc2833,
and INFO.
8. Can Reinvite: This feature is supported only
in the SIP protocol, and allows for two phones
to pass the voice traffic directly with each
other without going through the edgeBOX
after a few seconds.
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This is where an administrator would select
which audio codecs to use. Notice the G.729 is
grayed out, because a license must be
purchased from Digium in order to use them. We
recommend only allowing the codecs that the
phone supports as some equipment may have
trouble negotiating the best codec to use. Notice:
That you have to purchase a license from Digium
in order to use the G.729 codec.
The administrator may add groups of IPs where
the phones may not register.
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2 Registering Phones on the EdgeBOX
Until now we have only gone over creating the accounts. We will now go into how to use these
accounts. We will discuss two common devices registered on the edgeBOX one is a regular SIP
enabled phone and the other is a soft phone. I will also discuss where on a network these clients
can be placed. We will discuss three common places the edgeBOXes LAN, EWAN, and WAN.
Places on the network
In general all clients connected directly to the LAN, EWAN, or WAN will function right away. If
however a phone is connected to a different network say a private network on the Internet, you will
have to enable NAT in the advanced settings discussed in section 1.3.
2.1 Soft Phone configuration
We will start with configuration of a soft phone on the LAN of the edgeBOX using X-Lite a freeware
Softphone. Keep in mind that you may use any standard SIP or IAX2 compliant soft phone or VoIP
phone.
Open X-Lite and right click on it, choose “SIP Account Settings…”
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Click on add.
I have filled in all the fields here,
but most soft phones and VoIP
phones only need the following
information:
1. Username
2. Password
3. VoIP server Address
Keep in mind that they may be
called something else (as in the
next example), but essentially any
VoIP phone requires these 3 pieces
of information baring settings like
NAT, STUN or outbound proxy
settings they should register right
away if these three items are
properly configured.
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2.2 VoIP Phone configuration
The configuration of a VoIP phone is generally the same; there are really only three fields we
should be concerned about. They are highlighted in the screen shot below. NOTE: The names of
the fields are slightly different. In most cases you need only point your browser to the IP address of
the Phone to configure the settings.
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3 The LCR and outgoing calls
The LCR (Least Cost Routing) is used to configure the edgeBOX on how to route outgoing calls.
1. Route Name: This field is simply a name given to the route. Attention you may not
delete the Default route.
2. Inbound Pattern: This is the pattern used to differentiate the type of calls.
3. Hours: This allows the edgeBOX to use different routes based on time.
4. Type of Call: This assigns the type of call this route belongs too.
5. Routes: This field describes the route the call can take.
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To create a new outbound route simply click on “New Route”.
1. Inbound Pattern: This is the pattern used to differentiate the type of calls.
2. Route Name: This field is simply a name given to the route. Attention you may not
delete the Default route.
3. Type of Call: This assigns the type of call this route belongs too.
4. Hours: This allows the edgeBOX to use different routes based on time.
5. Routes: This field describes the route the call can take.
6. Outbound Pattern: This is where we can control the numbering pattern sent by the
edgeBOX.
7. Timeout: This is used to define an amount of time the edgeBOX try a specific route
before moving onto the next route.
8. CallerID: You can define a specific DID or name for each outbound route. This will
overwrite the all other callerID's from the system.
Example LCR:
Scenario:
One corporate headquarters (Portugal) and 2 remote offices(one in London and the other
in the United States).
VoIP needs:
1. All users must be able to dial each others extensions easily.
Most users are allowed only access to make locals. Mobile and long distance calls
can only be made by management.
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Solution:
Above we have a sample LCR. The LCR will automatically select the route based on what the user
dials using by selecting the best match. Mobile calls in Portugal all start with 9 if a normal user tries
to dial a number that starts with 9 the call will be rejected as they do not have permissions to make
that call. I have created an extension mapping scheme where all extensions that start with 3 are US
extensions and all extensions that start with 2 are UK extensions. Any call trying to dial a US
phone number (001 prefix) will be sent to the US office before going out to the PSTN. The three
offices are connected via remote switch (described later in this chapter section 3,2). notice that
international calls outside the US and UK are being sent to a VoIP provider with and ISDN backup.
This backup can be set up for all routes.
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3.1 VoIP Providers, SIP Proxies, and Soft Switches
The edgeBOX allows for connecting to various predefined VoIP providers. It will allow connections
to SIP Proxies and Soft Switches as well. In essence VoIP providers and Soft Switches all act as a
SIP Proxy. Lets discuss configuring a SIP proxy:
In VoIP and IPPBX setting select the outgoing calls tab from the list on the top, then the SIP proxy
tab from the list on the right. You can click on add new proxy to get the following window.
Proxy name is simply an identifier you may name as you like. The host field should be either the IP
address or the host name of the Proxy, VoIP provider, or Soft Switch. You can define the maximum
number of calls that can be carried simultaneously in MAX Calls. The callerID field can be
populated with the callerID information required by the Proxy. If you enable Trusted Proxy incoming
calls from this proxy will go directly to the internal call system. Keep alive will keep the connection
active. You can also select which codecs this connection can use.
After clicking on Advanced you will see the following options:
These options differ from proxy to proxy, and should be investigated. Most proxies which require
authentication will only require the username and password, however some do require other fields.
Enable Authentication:
1. Username: The username required by the Proxy to register.
2. Password: The password required by the Proxy to register.
3. Register Name: This is normally the same as the username sometimes it is
username@domain.
4. Authentication Name: This is normally the same as the username sometimes it is
username@domain.
5. From User: This is the username.
6. From Domain: This is the domain.
7. Realm: The realm may also be required, and is either the domain or a word that is
preset by the Proxy.
8. Contact: Contact can be the username, username@domain, or DID.
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This window allows you to configure NAT for the proxy.
Nat should only be used in the event that the other proxy or the edgeBOX are not on the same
network. The fields are self explanitory.
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3.2 Remote Switches
A remote switch works like a SIP Proxy, however it does not use SIP is uses the AIX2 protocol. This
protocol is useful for connecting edgeBOXes for two main reasons. It is simpler to set up and it
creates an IAX2 trunk which is more efficient in handling more than two calls at a time.
1. Name: the name you are giving this connection (Must be the same on both sides).
2. Secret: This is like a password used to authenticate the connection on both sides. (both
sides of the connection must use the same secret)
3. Host: This is the IP address of the other side.
4. You can also allow incoming calls and choose the codecs used in this trunk.
3.3 Calls to ISDN
The edgeBOX will automatically load certified and supported ISDN cards. The configuration of
these cards is straight forward.
1. You can change the mode between NT and TE mode, have the line in PMP(point
multi-point) or P2P (point to point).
2. Can allow Overlap Dial and make the line a Trusted Line.
3. You can also and MSN Numbers.
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3.4 Calls to Primary cards E1/T1
The edgeBOX will automatically detect supported PRI cards. These setting below should be
configured with your Primary line provider's settings in mind, or the Legacy PBX's settings in mind.
3.5 Analog Lines
Analog Lines are divided into two types FXO and FXS ports. FXO ports allow you to connect to the
phone carrier to make outgoing calls. While FXS ports are used to connect an analog phone
directly to the edgeBOX.
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3.6 Authentication
You can have the edgeBOX authenticate all non-free calls. Simply turn the authentication on in the
window below. Additionally you can add a prefix to gain access to outgoing calls.
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4 Incoming Calls
Incoming calls will go to one of two areas the IVR (interactive voice response) or the internal phone
system. We will start by discussing the IVR then move onto the internal phone system.
4.1 The IVR
The IVR is a reservoir for calls coming from outside the edgeBOX IP-PBX. You can access the IVR
via the window below. The IVR is organized by contexts Distinguished by the “green arrow” all other
icons represent actions belonging to contexts.
When editing the context, you will be given the following options:
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In the window above you can add, remove, and reorder actions for the context. You will can add the
following actions:
Trigger Conditions There are four triggers you can use to set off one of the actions below.
After Press: After the caller presses a specific number.
On start: After this context starts.
On timeout: After a given timeout has been reached.
Invalid Entry: If the caller enters an invalid number.
Actions
Dial: Dials an extension that can be configured.
Dial Out: This dials a number outside of the edgeBOXes IPPBX
Goto: Sends a call to another context.
Hangup: This action hangs up the call.
Voicemail: Sends the call to the voicemail box of your choice.
Queue: Sends a call to an existing queue.
Meet Me: Adds the call to a conference. You may choose any static conference previously
configured in the system.
Hunt Group: Sends the call to an existing hunt group.
Answer: Answers the call.
Background: Plays the sound files configured in the background this allows for the caller to
make choices as he is listening to the options.
Playback: This will play the sound files configured, but the caller has to wait until the files
have stopped playing before making choices
Wait: Tells the IVR system to way a configurable amount of time before moving onto the
next action.
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Example
You are asked to press 0 to speak to the Operator, press 1 for Sales
(which is a Sales hunt group), or dial the extension of the person you want to speak to.
The configuration will look like that of the image below.
4.2 Internal
The internal area of incoming calls allows for administrators to change the behavior of each
extension. Much like the IVR the administrator can add actions to each extension number. For
example an extension like 1000 can be sent to a hunt group then a voicemail account. as shown
below.
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4.3 Call Rules
Call Rules allow administrators to configure different IVRs depending on the time of day, day of the
week or month. An administrator can for example have incoming calls go to voicemail after
receiving a prerecorded message stating the office is closed between the hours of 8PM and 8AM.
In order to achieve this you will need to create two rules one from the hours 00:00 to 08:00 and
another from 20:00 to 23:59, as shown below.
4.4 DID Routes
Call Routes allow you to do the same thing as call rules, with the control being the line the call is
coming in on, and not time. For example you can have a number 555-1234 go to the IVR and 5555512 go directly to a specific extension.
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4.5 DID Ranges
DID Ranges allow you to map a group of external phone numbers to extensions. for example if you
have a primary line with 10 phone numbers (555-5550 to 555-5560) and would like them mapped to
extensions on your edgeBOX (200 to 210). You would fill in the DID Ranges window as follows.
4.6 Sound Manager
The sound manager allows an admin to upload custom sound files to the edgeBOX. These files
can then be used in IVR type menus.
5 The PBX
The PBX features included with the edgeBOX include agents, queues, hunt groups, conferences,
Parking and Voicemail.
5.1 Creating agents
Agents man the queues created to answer your customers needs, as seen below you can configure
the agents in a variety of ways.
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An agent can either be logged into a queue the entire time he is present, or the queuing system can
call him back every time it is his/her time to receive a call. To enable this feature an administrator
must check enable below “Call Back Login” and enter an extension used for the agents to log into.
The system can also be set to log off an agent if a call isn't answered in a given amount of time this
is called the “auto log off time”. The “require ACK” feature enables or disables an acknowledgment
key “#” to be pressed to answer a call. An admin will also be able to configure the music the agents
listen to when awaiting the next call.
Agent ID: This is the agents ID number.
PIN: A unique number used to authenticate an agent.
Agent Name: This is the agents name.
Login Extension: This is an extension number the agents would dial into to log in/out.
5.2 Creating queues
Queues are simple to create the require minimal settings configurations, as follows.
Queue Name: name assigned to this queue;
Extension: internal extension associated with this queue;
Announce Position Frequency: time interval (in seconds) between queue position
announcements;
Ring Strategy: algorithm used to assign calls to agents:
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RingAll: all agent phones will ring, and the call will be assigned to the one that answers
first;
RoundRobin: selects each agent in turn;
LeastRecent: selects the agent which was least recently selected;
FewestCalls: selects the agent with least calls answered;
Random: selects an agent randomly;
RR with Memory: RoundRobin with memory, remembers which agent answered last and
selects the next one.
Max Callers: maximum number of calls that can be placed on this queue;
Queue Priority: queue's relative priority to other queues configured;
Music on Hold: music that will be played when the call is queued;
Announce Hold Time: set to Yes if you want queue position to be announced, set to No
otherwise;
Leave When Empty: set to Yes if you want calls queued to be terminated if there are no
agents assigned to the queue.
5.3 Huntgroups
Hunt groups allow many phones to ring from only one call. The first phone to answer keep the call.
Simply choose a name and add the phones you want to ring.
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5.4 Confrences, Parking and Voicemail
●
Confrences
Confrences can either be static or dynamic if you would like to allow them. Dynamic confrences are
allowed if there is a check mark next to “Users Can Create Confrences”. For a user to create a
confrence he/she simple needs to dial the confrence extension and follow the instructions. A static
confrence simply click on “Add Room” and fill in the information required.
Conference Number: internal extension assigned to this conference;
Max Members: Max number of members in the conference
Protected Conference: If you enable this option, you will need to supply a conference PIN
and an administrator PIN. Users will then have to enter the correct PIN to join this
conference.
● Parking
The parking call service is ideal for transferring calls. To configure this service, you will have to
supply the following elements:
Number to Dial: number to dial to place the call on hold;
Number of Lines: number of extensions available for parking and
Parking Max Time: the maximum amount of time a call can remain on hold.
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To park a call, dial "#" plus the number configured in the "Number to Dial" field. A message will
inform you in which extension the call was parked into (let's assume it was 701). This call can then
be answered from any internal extension dialing 701.
● Voicemail
In the voicemail configuration panel you can define some of its functional parameters.
In general settings you can define:
Voicemail Extension: extension number where you can access the voicemail system and
hear your messages.
Max Messages: maximum number of messages that a user can have in his/her mailbox.
Max Length: Max length of message
Min Length: Min length of message
You can also define parameters to the notification messages, ie messages edgeBOX sends when
a user receives a new voicemail message:
From Email: origin e-mail address of notification messages.
From String: name of the entity originating notification messages.
Attach Message: when active, the voicemail message is attached to the notification
message in audio format.
Body Message Language: language used in notification messages. There are two available
options: English and Portuguese.
Signature: signature of the notification messages.
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6 Resources
6.1 CDR Files
CDR Files are used to store call detail records. They are created automatically and can be
downloaded via FTP to a your computer for inspection, or to be used with billing software. The file
is in csv format. In order to access the file via ftp there are some steps to be performed first.
Step1: Create a logmaster password. This is done through the GUI by going to the following
'Syestem » Logging » Logs'.
Step2: Ftp to the edgeBOX as logmaster and select the file to download.
The files are stored with a filename of Master.csv (the current log file). The log files are rotated daily
(Master.csv.1-7) and kept for 7 days (after which the oldest file is overwritten by the new log file).
The entries in the Log file have the following meaning:
1. accountcode: What account number to use (Only used when Authentication is enable)
2. src: Caller*ID number
3. dst: Destination extension
4. dcontext: Destination context
5. clid: Caller*ID with text
6. channel: Channel used
7. dstchannel: Destination channel if appropriate
8. lastapp: Last application if appropriate
9. lastdata: Last application data (arguments)
10. start: Start of call (date/time)
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11. answer: Anwer of call (date/time)
12. end: End of call (date/time)
13. duration: Total time in system, in seconds (integer)
14. billsec: Total time call is up, in seconds (integer)
15. disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY
16. amaflags: What flags to use: see amaflags::DOCUMENTATION, BILL, IGNORE
6.2 Sites
The edgeBOX Manual
http://www.voip-info.org/wiki/
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