MT 5.3 Settings

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Settings

Settings: On master tenant

Settings: On master tenant

Master Settings are used to set global MT variables. These settings are applied on the entire system if not otherwise specified on a lower level.

For example: 'Settings: Voicemail: Send Attachment'='Yes' will set 'Set Attachment' for all local and remote extensions. But each extension can override this rule by setting 'Extensions: Edit: Advanced Options: Attach'='No'



Tenants

PBXware MT system administration theoretically allows up to 800 slave tenants to be administered from a single administration interface, but more than 100 is not recommended.

Tenants
  • Tenant Name:
Unique, custom set, tenant name.
(ex. Slave Tenant #2)
(Display)
  • Tenant Code:
Unique custom set tenant code.
(ex. 300)
  • Package:
Name of the package used by a slave tenant.
(ex. Full package)
(Display)
  • 5.0.edit.png Edit
Edits the tenant configuration
(ex. Click to edit tenant configuration)
(Button)
  • 5.0.delete.png Delete
Deletes a tenant from the system
(ex. Click to delete a tenant from the system)
(Button)


Network Info

Network Info
Network Info
  • Server Name:
Set the master or slave tenant name here.
(ex. Slave #1)
([a-z][0-9])
  • Jabber server ID:
ID on which Jabber users will log on.
(ex. voip.domain)
([a-z][0-9])
  • gloCOM SIP Proxy (only on master tenant):
Public IP address/domain of gloCOM SIP Proxy. In case you are using SIP proxy for registration to your tenant set its address here and gloCOM will use value from this field for registrations.
ex. voip.domain.com or 192.168.1.59
  • Packages:
Tenant package that the slave is using:
(ex. Full package)
(Select box)
  • Unique Tenant Code:
This is the unique code assigned to a slave tenant. When a phone is registering to an extension, its username is set to TENANT_CODE+EXT_NUMBER. You can specify only numbers between 200 and 999.
(ex. 200)
([0-9])

General Settings

Master Tenant Settings

Master Tenant

  • Default Server:
Which tenant is shown in the interface by default.
(ex. Select Yes to make a tenant default when showing the interface, which will set all other tenants to No).
(Option buttons)
  • Announce Trunks:
Announce over which trunk the call goes through
(ex. John dials 55510205 and this call goes over the secondary default system trunk. John will hear 'Using the secondary trunk to terminate your call' message).
(Option buttons)
  • Absolute Timeout:
Maximum time a call can last (in seconds)
(ex. If '3600' is set in this field, that will make all calls end after 1 hour ( 1h = 3600 seconds))
([0-9])
  • Voicemail in CDRs:
Sets how the calls that were unanswered and redirected to voicemail are displayed in CDR
(ex. A call was made to extension 1000 but was not answered. The caller gets redirected to voicemail. If 'As Not Answered call' is set, CDR will display this voicemail redirection as an 'Unanswered' call. If no option is set, the same call will be displayed as 'Answered')
([0-9])
  • Fax page format:
Page format when downloading fax as a PDF. Available choices: letter, legal, A4 and Auto.
(ex. letter)
(Select box)
  • Fax file type:
Type of file for downloading. Available choices: Both PDF and TIFF, Only PDF, Only Tiff.
(Select box)
  • Routing mode:
What type of routing mode will be used for Routes. (Only on master tenant)
Example:
  • E.164 routing - routing mode where routes are set using E.164 numbering rules
  • Simple routing - routing mode where you manually set routes to fit your needs
(Select box)
  • DID Mode:
(ex. Standard, Groups)
  • Default domain for OSC login:
Default domain used for OSC login.
(ex. http://pbx.domain.com/)
([a-z][0-9])
  • Enable Tenant to Tenant calls:
When this option is enabled, extensions from one tenant will be able to call extension from another tenant which is done dialing TENANT+EXT
(ex. Yes, No, N/A)
(Option buttons)
  • Disable CallerID rewrite for tenant to tenant calls:
If enabled, any call between tenants will not have overwritten CallerID. It will send anything that extension has set on it.
(ex. Yes, No, N/A)
(Option buttons)
  • Find E.164 numbers in DIDs:
Enable or disable E.164 matching of numbers as set in the E.164 field in DIDs
(ex. Yes, No, N/A)
(Option buttons)
  • Number of objects per page:
Number of objects per page like extensions, DIDs and trunks.
(ex. 20)
([0-9])
  • Show Directory in OSC:
Whether to show the Directory in Online Self Care. Only those extensions which allow this, will be shown in Directory
(ex. Yes, No, N/A)
(Option buttons)


Tenants

Tenant Settings
  • Default Server:
Which tenant is shown in the interface by default.
(ex. Select Yes to make a tenant default when showing the interface, which will set all other tenants to No).
(Option buttons)
  • Announce Trunks:
Announce over which trunk the call goes through
(ex. John dials 55510205 and this call goes over the secondary default system trunk. John will hear 'Using the secondary trunk to terminate your call' message).
(Option buttons)
  • Absolute Timeout:
Maximum time a call can last (in seconds)
(ex. If '3600' is set in this field, that will make all calls end after 1 hour ( 1h = 3600 seconds))
([0-9])
  • Voicemail in CDRs:
Sets how the calls that were unanswered and redirected to voicemail are displayed in CDR
(ex. A call was made to extension 1000 but was not answered. The caller gets redirected to voicemail. If 'As Not Answered call' is set, CDR will display this voicemail redirection as an 'Unanswered' call. If no option is set, the same call will be displayed as 'Answered')
([0-9])
  • Fax page format:
Page format when downloading fax as a PDF. Available choices: letter, legal, A4 and Auto.
(ex. letter)
(Select box)
  • Fax file type:
Type of file for downloading. Available choices: Both PDF and TIFF, Only PDF, Only Tiff.
(Select box)
  • Enable Tenant to Tenant calls:
When this option is enabled, extensions from one tenant will be able to call extension from another tenant which is done dialing TENANT+EXT
(ex. Yes, No, N/A)
(Option buttons)
  • Tenant groups:
Selected groups will be able to make calls between tenants within that group.
(ex. Select Tenant group to enable Tenant to Tenant calls)
(dropdown)
  • Disable CallerID rewrite for tenant to tenant calls:
If enabled, any call between tenants will not have overwritten CallerID. It will send anything that extension has set on it.
(ex. Yes, No, N/A)
(Option buttons)
  • Default CallerID:
If set, calls that are going out through a trunk assigned to this tenant will have this CallerID. (Only on slave tenants)
(ex. Tenant-001)
([0-9][a-z])
  • Use Default CallerID for tenant to tenant calls:
Whether the Default CallerID will be used when making calls from the extension on one tenant to extension on another tenant. (Only on slave tenants)
(ex. Select Yes to use Default CallerID).
(Option buttons)
  • Use DIDs as CallerID for tenant to tenant calls:
This option will allow administrator to keep the default behavior in place but also allow the option to disable it if necessary.
(ex. Yes, No, N/A)
(Option buttons)
  • Find E.164 numbers in DIDs:
Enable or disable E.164 matching of numbers as set in the E.164 field in DIDs
(ex. Yes, No, N/A)
(Option buttons)
  • Number of objects per page:
Number of objects per page like extensions, DIDs and trunks.
(ex. 20)
([0-9])
  • Show Directory in OSC:
Whether to show the Directory in Online Self Care. Only those extensions which allow this, will be shown in Directory
(ex. Yes, No, N/A)
(Option buttons)

Operation Times

Set the tenant's open/closed times. Depending on the time when the call is received, the call can be redirected to different PBXware destinations.

System Operation Times
  • Operation Times
Enable operation times
(ex. Yes, No)
(Option buttons)
  • Default Destination
PBXware extension all calls are redirected to during the closed time hours
(ex. 1000)
([0-9])
  • Greeting
Greeting sound file played to callers during the closed times
(ex. greeting-***)
(Select box)
Description of destinations follows in this priority order:
  • Open dates: Sets the working hours during which DID is to redirect calls as set in DID Add/Edit window. If any call is received during the hours not set here, 'Custom Destination' are checked, and if they do not apply, the call is redirected to 'Default Destination' (Closed dates)
  • Custom Destinations: Redirects all calls received during set hours to PBXware extension provided here
  • Closed dates: Sets the specific date/s when all calls are redirected to the 'Default Destination'. If the 'Destination' field in the Closed dates is set, calls will not go to the 'Default Destination' but to this number.
NOTE: If you manually open/close system using *401/*402 access codes, the time rules that you have defined in Operation times will not be valid for that day. After midnight they will be applied again to the system.
Operation Times closed dates

Call Recordings

Call Recordings
  • Record calls by default:
Setting this option to Yes will enable call recordings on entire system.
(ex. Yes, No or N/A)
(Option buttons)
  • Silent recording by default:
If "Record calls by default" is set to "Yes", enabling this option will play message "Recorded" to users before call is connected, informing them that call will be recorded.
(ex. Yes, No or N/A)
  • Auto MP3 Conversion:
Convert files to mp3, this will apply to system-wide recordings
Select options from the drop-down:
  • Convert and remove original
Will convert all files to MP3 and erase original files.
  • Convert and keep original
Will convert all files to MP3 and keep original files. (recordings can't be played from GUI).
  • Convert on Listen/Download
Recordings are left in their original format and only converted to mp3 when requested for listening/download, leaving behind both the original and the mp3 version of the recording after listen.
  • Do not convert
Will leave files in the original format without conversion. (recordings can't be played from GUI).
(drop-down)
  • Email instant recording interval (min):
Instant recordings are not e-mailed immediately after they are recorded, but rather periodically. This field will set how often PBXware will send recordings out.
(ex. 3)
([0-9])
  • Recordings format:
Format used for saving the system call and voicemail recordings. You can read more details about disk space usage on the bottom of this chapter
(ex. Choose one of the following formats: gsm, wav, wav49 and ogg. If wav is selected, all call recordings and voicemail recordings will be save in this format).
(Select box)
  • Use RAM disk:
To speed up the process of recording you can turn on the RAM disk which practically records all calls to RAM memory before storing them on the disk
(ex. Yes, No, N/A)
(Option buttons)
  • RAM disk size:
Amount of memory used for call recording
(ex. Yes, No, N/A)
(Select box)


Disk Space Used By Call Recording:

With continuous tone for 60 seconds:
  • wav49 = 84.5kb
  • wav = 833.0kb
  • gsm = 85.0kb

With continuous silent tone (without sound) for 60 seconds:

  • wav49 = 84.0kb
  • wav = 827.0kb
  • gsm = 84.0kb


Disk Space Used By Voicemail Recording

With continuous tone for 60 seconds:
  • wav49 = 91.0kb
  • wav = 863.0kb
  • gsm = 91.0kb

With continuous silent tone (without sound) for 60 seconds:

  • wav49 = 0.38kb
  • wav = 3.0kb
  • gsm = 0.32kb


Localization

Localization
  • Voicemail Localization:
This setting affects the language used for the datetime string provided by voicemail. Field should be set to a valid locale available on the system (i.e. en_US.UTF-8).
Anything that is used in voicemail datetime template that is not part of the date will not be localized. Example: 'Tuesday, March 21, 2017 at 09:20:59 AM' here 'Tuesday' and 'March' would be localized while 'at' would remain unchanged regardless of the selected localization. Additional variables are available for E-mail templates:
%VM_LOCAL_DATE%: Contains the date part in format 'weekday, month day, year' eg: ' Friday, March 24, 2017'. Weekday and month are localized according to selected locale.
%VM_TIME%: Contains the time formatted according to the selected locale eg: en_US locale will set VM_TIME in 12 h format while de_DE locale will set it in 24 h format.
Existing %VM_DATE% variable will be modified to remove the 'to' from it if a locale is set for that tenant.
  • Audio Language:
This changes the language in which the sound files are played.
NOTE: Files will be played in another language only if the sound files for that language exist in Asterisk.

Features

Features
  • Hide CallerID in OSC:
Whether to hide CallerID in Online Self Care
(ex. Yes, No, N/A)
(Option buttons)
  • Force "Allow ES CallerID" in Call Forwarding:
Force this option when in the Call Forwarding part of Enhanced Services
(ex. Yes, No, N/A)
(Option buttons)
  • Ringtone for Local calls:
Default Custom Ringtone for Local calls.
(ex. <Simple-3>)
([a-z][0-9])
  • Set CallerID for Group Hunt calls:
If numbers set in Group Hunt have one or more remote numbers, those numbers will receive a different Caller ID on the call. That Caller ID is the one from the original caller set in its 'Caller ID' enhanced service for the trunk where the call was going through
(ex. Yes, No, N/A)
(Option buttons)
  • Only Allow Trunk CallerID within DID range:
If the calls from some other server are going through this system's Multi User extension, and CallerIDs from other server are matching DIDs on this server, they are properly shown to callees. Otherwise CallerID is represented as Anonymous
(ex. Yes, No, N/A)
(Option buttons)
  • On DID save update ES/CID/Trunks:
Whether to set CallerID on the extension's enhanced services to a value which is the same as the DID number
(ex. When this option is turned on and you save a DID it will set its number as a CallerID of the extension to which it points. This will happen when DID is pointing to said extension, and the CallerID in enhanced services of the extension will be set for the same trunk that is selected in DID).
(Option buttons)
  • Do not allow users sending any CallerID:
If enabled, system will not allow extensions/users sending any CallerID.
NOTE: this option will work only if CallerID is not set on Tenant, Extension or DID.
(ex. Yes, No, N/A)
(Option buttons)
  • Hide Extensions with no department (gloCOM):
If Extension doesn't have department assigned, for example, Department is set to None, then this Extension will not be shown in gloCOM.
(Yes/No/NA)
  • Disable local DID matching:
If turned 'Yes', the system will disable local DID search and will, in turn, go through trunk to search a match on another system.
(Yes/No/N/A)
DID matching
  • Use Dynamic Features:
By using dynamic features you can send a DTMF to activate specific dialplan. For example, when this option is set to No, you will not be able to pause recordings, and you will be able to use direct media.
(Yes/No/N/A)
  • Disable Call Rating for Call Forwarding:
(ex. Yes, No, Not Set)
  • Allow IP Address Authentication for Extensions:
Allow IP Address Authentication for Extensions and force IP based authentication.
(ex. Yes, No, Not Set)
IP Address authentication
When this field is set the following fields are enabled per Extension:
1.Incoming IP Addresses
2.Insecure
  • Skip Voicemail PIN Prompt:
Choose whether the tenant/server users have to type in their PIN when entering voicemail options. Setting this option to Yes will skip the pin entry when dialing *123 (*124 should work as before with the pin).
(Option buttons)

Other Networks

The system can be part of the 'default' PBXware network where all extensions share the same unified dial plan. This is achieved by selecting the PBXware network that the system belongs to from the select box.

Clicking on 'Other network' will open the following options:

Other Networks
  • Mode:
Sets the way other PBXware networks are dialed
Example:
  • With Access Code - Access code + network number + extension (e.g. *188 8 1000)
  • Without Access Code - network number only + extension (e.g. 8 1000)
(Option buttons)
  • Name:
Other network name
(Network London)
(Display)
  • Number:
Other network access number
(ex. 8)
(Display)
  • Trunk:
Trunk used once the other network number is dialed
(ex. 2554433)
(Display)
  • 5.0.edit.png Edits the other network configuration
(ex. Click to edit the other network configuration)
(Button)
  • 5.0.delete.png Deletes other network connections
(ex. Click to delete other network connections from the system)
(Button)

In order for users to use prefixes that are 2 or 3 digits long you will have to enable "Mode: * With Access Code" 2 or 3-digit dial prefix. After you set this up, you must dial *188 XXX YYYYYYY, where XXX is 3-digit prefix and YYYYYY is the phone number you want to dial.

In case you have "Mode: * Without Access Code" (1 digit dialing) enabled, users will be able to dial the numbers that are set with your single digit prefixes without using *188 access code, simply typing X YYYYYYY, where X is your 1-digit prefix and YYYYYYY is number you would like to dial.

Add/Edit Network
Other Networks
  • Name:
Custom Other Network name/identifier
(ex. London FO/7)
([a-z][0-9])
  • Prefix:
Number used to access Other Networks (Up to 3 digits allowed)
(ex. If this field is set to '7', dial '*188 7 {NETWORK NUMBER}')
([0-9])
  • Strip Prefix:
Should the 'Prefix' number be stripped once dialing the Other network
(ex. If the 'Prefix' field is set to '7' and this field is enabled, once the user dials *188 7 55510205 the system will dial 55510205. If this field is disabled, 755510205 will be dialed).
(Option buttons)
  • Allowed Range:
Number range allowed to be dialed after the *188 Other Network Access Code

NOTE: The type of an entry in the Allowed Range field must be a regular expression. Let's say we want to enter '7' as our Prefix number. In this case the correct input with prefix 7 is: [0-9]{3}. The first part of an expression [0-9] represents one digit from 0 to 9. And {3} represents quantity (3 digits from 0 to 9). The correct solution is also: [0-9][0-9][0-9]. It’s important to mention that the prefix number has to be excluded from the regular expression, since it’s already included by entering a number in the Prefix field.

(ex. If this field is set to '2', only extensions on the Other Network starting with the number 2 will be allowed to dial. If you dial *188 7 1002, your call will fail. But, if you dial *188 7 2000, your call will be transferred to extension 2000).
([a-z][0-9])
  • Hidden Prefix:
Prefix number added before dialed number
(ex. If this field is set to 212 and we dialed *188 7 55510205, 21255510205 will be dialed. This is useful if the provider requires a certain code before dialing the destination, area code inserted automatically, etc.)
([a-z][0-9])
  • Trunk:
Select the trunk that will be used once 'Number' is dialed
(ex. London FO Trunk)
(Select box)


Speed Dial

Speed Dial is used with the *130 Access Code. When you dial *130XX, where XX is a two digit Speed Dial Code, you will dial the extension associated with that code.

The difference here is that you set Speed Dial Codes for all extensions on the system, not just the ones which have this turned on in ES

TIP

This is useful only if you have more than 6 digits in your extensions.

Speed Dial
  • Code (XXX)
Three digit code which is entered after the Speed Dial Access Code, *130 as default
(ex. 22)
([0-9])
  • Speed Dial Name
Short description of the Destination to which this Code points.
(ex. Sales-John)
([a-z][0-9])
  • Destination
Destination to which this Code is pointing.
(ex. 1005)
([0-9])

CSV Upload is used when you have all the codes written in a simple CSV file in form:

  • Code,Name,Destination

CSV Download is used when you want to download already set Dial Codes in CSV file

Administration

This section enables remote administration of the system and is set on master tenant.

Administration
  • Daemon username:
Daemon username
(ex. admin)
([a-z][0-9])
  • Daemon passwd:
Daemon password
(ex. GTaXfgtR)
([a-z][0-9])
  • Port:
PBXware MT connection port
(ex. 10001)
([0-9])
  • AGI Port:
Agi connection port
(ex. 4573)
([0-9])
  • Allow LDAP username:
This tells PBXware MT to accept registration from GLOCOM when it is using a Windows® username for this.
(ex. Yes, No, N/A)
(Option buttons)

Email

System during its operations sends email notifications and alerts to various users and administrators. These emails can be sent using a built-in 'local mail server' or remote SMTP server.

Email
  • From E-mail:
What email to show in the From: field.
(ex. info@domain.com)
([a-z][0-9])
  • Voicemail From E-mail:


A From E-mail is used to set the From: field in emails that are sent. E-mail authentication and few additional options can be configured after you go to the Setup Wizard at https://IPADDRESS:81 and click on SMTP Configuration button.

SMTP Configuration
  • E-mail Account:
Address to which email will go if the recipient is not specifically defined
(ex. john@domain.com)
([0-9][a-z] @)
  • SMTP Address:
The host to send mail to, in the form "host | IP_addr"
(ex. mail.domain.com)
([0-9][a-z])
  • SMTP port:
Port used to send emails to the host
(ex. Default port is 25)
([0-9])
  • Authentication:
Select whether authentication with the SMTP server is needed or not
(ex. On)
(Option buttons)
  • Username:
Username used for SMTP AUTH
(ex. username)
([a-z][0-9])
  • Password:
Password used for SMTP AUTH
(ex. password)
([a-z][0-9])
  • Allow From Override:
Are users allowed to set their own From: address
Example:
YES - Allow the user to specify their own From: address
NO - Use the system generated From: address
(Option buttons)
  • Use SSL:
Specifies whether ssmtp uses TLS to talk to the SMTP server
(ex. The default is NO)
(Option buttons)
  • Use Start TLS:
Specifies whether ssmtp does a EHLO/STARTTLS before starting SSL negotiation
(ex. See RFC 2487)
(Option buttons)

Locality

Locality sets from where the system is operating, and it is set for every slave in particular.

Locality
  • Country:
Country from which the tenant is operated
(ex. If the tenant is operated from the USA, set USA here)
(Select box)
  • Tenant Time Zone:
Setup a default timezone for a tenant that will apply to all extensions on the target tenant.
(ex. If the system is located in the USA, set the USA here, otherwise, select the closest country to yours)
(Select box)
NOTE: Only clean installation and MT license will have timezone option enabled in (first Setup Wizard). If you relicense, it is considered like 'old' system. For old systems script needs to be used to make the conversion (to GMT).


  • Voicemail Time Zone:
Setup a default timezone for a voicemail that will apply to all extensions on the target tenant.
(ex. If the system is located in the USA, set the USA here, otherwise, select the closest country to yours)
(Select box)
  • Area Code:
Area code from which the tenant is operating
(ex. New York area code - 212)
([0-9])
  • National Dialing Code:
Code needed for dialing national destinations
(ex. 1(USA), 0 (United Kingdom and Germany))
([0-9])
  • Leave National Code:
In some countries, the national code is stripped automatically. If set to 'Yes', the national code will not be stripped from the dialed number.
(ex. 035123456 will not be stripped of 0)
(Option buttons)
  • International Dialing Code:
Code needed for dialing international destinations
(ex. 011(USA), 00 (United Kingdom and Germany))
([0-9])
  • Emergency Services
When you click on the Emergency Services button, a pop-up window will allow you to enter Police, Fire, and Ambulance numbers.
Emergency Services - Notification E-mail.
After dialing one of the emergency numbers an e-mail will be sent to the provided address. If the e-mail field is left empty no e-mail will be sent.
Note: Setting this on Master Tenant will not affect anything.
Emergency Services
  • Currency symbol/ISO code:
  • Currency writing convention:
- Symbol before amount
- Symbol after amount

Auto Provisioning

Auto provisioning sets values to be used for the auto provisioning system in order to create auto provisioning files correctly depending on the UAD locality (local/remote). This is set on master for all tenants.

Auto Provisioning
Master tenant:
  • LAN Hostname/IP:
Local area network IP address used to auto provision local UADs
(ex. 192.168.1.2)
(IP address)
  • WAN Hostname/IP:
Wide area network IP address used to auto provision remote UADs
(ex. 192.168.1.2)
(IP address)
  • gloCOM LAN server
PBXware MT LAN IP address that users, connecting to PBXware from LAN, will use to register their gloCOM clients.
  • gloCOM WAN server
PBXware MT WAN IP address that users, connecting to PBXware from WAN, will use to register their gloCOM clients.
  • WAN Host for DNS SRV
WAN IP address/domain for DNS SRV host
ex. dnssrv.domain.com
  • Enable TFTP:
TFTP server is isabled by default for new installations set "Yes" to turn it on again
(ex. Yes, No, N/A)
(Option buttons)
  • Use DNS SRV when possible:
Whether Polycom phones should use DNS SRV records to find out SIP port from DNS server
(ex. Yes, No, N/A)
(Option buttons)
Slave tenant:
  • HTTP username
Username for HTTP auto provisioning.
  • HTTP password
Password for HTTP auto provisioning.

NOTE: PBXware MT 3.8.2 introduced our new dynamic auto provisioning system that supports both TFTP and HTTP provisioning simultaneously. However, in order to be able to use HTTP provisioning, your devices must support username and password with digest. For security reasons HTTP auto provisioning will be disabled until HTTP username and HTTP password are set up.

Channels

Depending on the CPU power of the server a custom number of channels can be assigned for various channel types

Channels
  • Local Channels:
Total number of all channels used by local UADs
(ex. 12)
([0-9])
  • Remote Channels:
Maximum number of active inbound or outbound channels for specific tenant.
(ex. 12)
([0-9])
  • Conferences:
Total number of all system conferences
(ex. 8)
([0-9])
  • Queues:
Total number of all system ACD queues
(ex. 8)
([0-9])
  • Auto Attendants:
Total number of all system IVRs
(ex. 8)
([0-9])
  • DAHDI:
Total number of all system trunks using DAHDI protocol
(ex. 8)
([0-9])

The System will limit the number of channels in order to achieve and maintain excellent calls and other services' quality

TIP:

Channels can be used as a way of suspending entire tenant without actually having to delete anything. Setting Local and Remote Channels values to 0 will disable calls on the Tenant.

Extensions channel limit notification

Here you can set whether there will be any notification when users exceed their outgoing limits on calls. These options apply to all extensions.

Limit Notification
  • Play Sound:
This option tells the system to play a notification sound when any extension exceeds its outgoing call limit.
(ex. Yes, No, N/A)
(Option buttons)
  • Send E-mail:
Whether or not to send a notification mail when exceeding the limit.
(ex. Yes, No, N/A)
(Option button)
  • Notification e-mail:
E-mail address to which notification mail should be sent to if the number of calls exceed the limit.
(ex. user@domain.com)
([a-z][0-9] @)

Default Codecs

Default codecs can be set for the following groups:

  • Local - Local extensions
  • Remote - Remote extensions and Trunks
  • Network - PBXware MT network(two or more servers)

NOTE: Default codecs for newly created tenants (5.3.5 update):

When creating a new tenant, the default codecs will be inherited from Master tenant. Before this change, the default codecs were hardcoded.

tip.png Tip

Once a local/remote extension or a network is added/edited, only the codecs allowed here will be available for the extension/network usage.


Default Codecs

Available Codecs:

  • ITU G.711 ulaw' - 64 Kbps, sample-based, used in US
  • ITU G.711 alaw - 64 Kbps, sample-based, used in Europe
  • ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
  • ITU G.726 - 16/24/32/40 Kbps
  • ITU G.729 - 8 Kbps, 10ms frame size
  • GSM - 13 Kbps (full rate), 20ms frame size
  • iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
  • Speex - 2.15 to 44.2 Kbps
  • LPC10 - 2.5 Kbps
  • H.261 Video - Used over ISDN lines with resolution of 352x288
  • H.263 Video - Low-bit rate encoding solution for video conferencing
  • H.263+ Video - Extension of H.263 that provides additional features that improve compression over packet switched networks.

Monitoring

Monitoring sets alarms and notifications at which the system will monitor itself for normal operation and where by appropriate notifications are sent if alarms are triggered.


tip.png Tip

Reloading the system will not interrupt any services while restarting the system does stop and starts all system services. This is set on master tenant and affects all other tenants.


Monitoring
  • Monitor (mins):
Time interval at which the system should check if Asterisk is down. If down, the system will try to start it and will send a notification email about the stop/start action
(ex. 15)
([0-9])
  • Reload Type:
Select whether to reload the system at some specific time of a day or in regular time intervals (hourly)
(ex. Setting this option to 'Time of the day' and 'Reload (hours)' = '2' will reload the system every day at 02:00 hours. Setting this option to 'Regular Interval' and 'Reload (hours)' = '2' will reload the system every two hours)
(Select box)
  • Reload (hours):
This field is active only when the 'Reload Type' option is selected
(ex. Setting 'Reload Type' = 'Time of the day' and this option to '2' will reload the system every day at 02:00 hours. Setting 'Reload Type' = 'Regular Interval' and this option to '2' will reload the system every two hours)
(Select box)
  • Restart Type:
Select whether to restart the system at some specific time of day or in regular time intervals (hourly)
(ex. Setting this option to 'Time of the day' and 'Restart (hours)' = '2' will restart the system every day at 02:00 hours. Setting this option to 'Regular Interval' and 'Restart (hours)' = '2' will restart the system every two hours)
(Select box)
  • Restart (hours):
This field is active only when the 'Restart Type' option is selected
(ex. Setting 'Restart Type' = 'Time of the day' and this option to '2' will restart the system every day at 02:00 hours. Setting 'Restart Type' = 'Regular Interval' and this option to '2' will restart the system every two hours)
(Select box)
  • Notification e-mail:
Email address on which reload/restart notification is sent
(ex. email@domain.com)
([a-z][0-9])

Tenant packages

Tenant packages are "presets" that are used when determining how many of the following, slaves can have: extensions, voicemails, queues, IVRs, conferences and ring groups.

Tenant packages
  • Package name
Name of the package.
(ex. Full package)
(Display)
  • Usage count
Number of tenants using this package.
(ex. 4)
(Display)
  • edit.png Edit
Edits the package configuration
(ex. Click to edit package configuration)
(Button)
  • delete.png Delete
Deletes the package
(ex. Click to delete the package)
(Button)


Add Package

Add Package
  • Name:
Name of the tenant package.
(ex. Medium package)
([0-9][a-z])
  • Extensions:
Number of extensions that can be created on a tenant using this package.
(ex. 10)
([0-9])
  • Voicemails:
Number of voicemails that can be created on a tenant using this package.
(ex. 6)
([0-9])
  • Queues:
Number of queues that can be created on a tenant using this package.
(ex. 2)
([0-9])
  • IVRs:
Number of IVRs that can be created on a tenant using this package.
(ex. 3)
([0-9])
  • Conferences:
Number of conferences that can be created on a tenant using this package.
(ex. 4)
([0-9])
  • Ring Groups:
Number of ring groups that can be created on a tenant using this package.
(ex. 2)
([0-9])
  • Hot Desking:
Number of hot desking devices that can be created on a tenant using this package.
(ex. 2)
([0-9])
  • Restrict Service Plans:
Restrict service plans usage use for this package.
(Option buttons)
  • Allowed Service Plans:
Allowed service plan use for this package.(Only if Restrict service plans is "YES")
(Select box)
  • Default Service Plan:
Select default service plan for tenant using this package.
  • Call Recordings:
Choose to enable or disable Call Recordings on tenant using this package. If you disable it, you will not be able to record call on extension's level and DID. Options like Instant Recording, Delete Recording and Listen to Recordings in Enhanced Services will be disabled for all extensions, in case that you have set Call Recordings to No.
  • Call Monitoring:
Enabling or disabling this option will allow or disallow using Call Monitoring option in Enhanced Services for all extensions.
  • Call Screening:
If you enable this option, Call Screening option under Enhanced Services will be listed. User will have ability to enable or disable this option. Setting Call Screening option to No, will not show this option in Enhanced Services.

Tenant Groups

Add Package

You can create one or more tenant to tenant calls groups with the list of tenants allowed to call each other.

Group name: Enter group name
(text box)
Tenants: select two or more tenants to enable the tenant to tenant calls
(Select box)

Protocols

Protocol is a set of rules that allows UAD, systems, networks, etc. to communicate using a set standard.

Supported protocols are:

  • SIP
  • IAX

NOTE: PBXware 5.1 does not support Woomera / BRI any more.


SIP

SIP (Session Initiated Protocol, or Session Initiation Protocol), is a signaling protocol for Internet conferencing, telephony, presence, events notification, and instant messaging. The protocol initiates call setup, routing, authentication, and other feature messages to end points within an IP domain.

General

General
  • Port:
Port for SIP server to bind to
(default value 5060,)
([0-9])
NOTE: If PBXware is operating behind NAT, and you would like to change values for Port and External IP you also have to set External TCP port.
  • Bind Address:
IP address for SIP server to bind to (0.0.0.0 binds to all interfaces)
(default value 0.0.0.0,)
([0-9])
  • Default Context:
Default context for incoming calls
(ex. For security reasons it is recommended to keep this field set at 'invalid-context')
([a-z][0-9])
  • Default Language:
Default language settings for all users/peers
(ex. Set this option to 'en' (English) for example)
([a-z])
  • Default MoH:
Set the default MOH (Music on Hold) class for all SIP calls
(ex. Set 'default', for example, to play 'default' MOH class to all SIP calls when placed on hold for example)
(Select box)
  • Default Parking lot:
Sets the default parking lot for call parking.
This may also be set for individual users/peers. Parkinglots are configured in features.conf
([1-9][a-z])
  • Allow SIP Transfers:
Disable all transfers (unless enabled in peers or users). Default is enabled.
(ex. Yes, No, N/A)
(Option buttons)
  • Async operations:


  • Allow transport reload:


  • Outbound proxy:
This option will set PBXware to send outbound signaling to this proxy, instead directly to the peer
ex. proxy.provider-domain.com
type ([a-z]1-9|)

TCP Settings

TCP Settings
  • Enable TCP:
Enable server for incoming TCP connections. (Default value: No)
ex. Yes, No, N/A
  • External TCP Port:
This option allows you to define external port number for incoming TCP connections.
Default port used for TCP connections is 5060. ([0-9])
  • TCP Keep-Alive Interval
This will send \r\n\r\n packets to TCP/TLS endpoints in order to prevent NAT from closing port due to inactivity.
Enter time in seconds after which \r\n\r\n packets will be sent. Default value is 20 seconds. ([0-9])

TLS Settings

TLS Settings
  • Enable TLS:
Enable server for incoming TLS (secure) connections.
Options: Yes, No, N/A (default is no).
  • TLS Port:
This port overrides any TCP/UDP settings in SIP protocol settings and applies for all users (external and internal).
Default port used for TCP connections is 5061. ([0-9])
  • External TLS Port:
This option allows you to define an external port number for incoming TLS connections.
Default port used for TCP connections is 5061. ([0-9])
  • Use TLS for gloCOM:
Set Yes to force gloCOM Mobile/Desktop to use SIP TLS.
Options: Yes, No, N/A.

NAT

NAT
  • External IP address (:port):
External IP/Public/Internet address system uses.
If your system is behind NAT, set this option to Public/Internet IP address system uses when registering with other proxies over the Internet. In case this field is not populated users will experience one way audio registering from WAN
([0-9])
  • External Hostname (:port):
DynDNS address system uses
If your system is behind NAT, along with the External IP address you may use the DynDNS service. Set this field to DynDNS host.
([0-9])
  • Local networks (seperated by newline):
If the system is used in local network, set the local network address here
(ex. 192.168.0.0/255.255.0.0)
  • NAT:
Global SIP NAT setting which affects all users/peers.
Options:
  • yes - forces RFC 3581 behavior and enables symmetric RTP support
  • no - only enables RFC 3581 behavior if the remote side requests it and disables symmetric RTP support
  • force_rport - forces RFC 3581 behavior and disables symmetric RTP support
  • comedia - enables RFC 3581 behavior if the remote side requests it and enables symmetric RTP support
(Option buttons)

Security

Security
  • Authenticate OPTIONS requests:
Enabling this option will authenticate OPTIONS requests just like INVITE requests are. By default this option is disabled.
Options: Yes, No, N/A

RTP Timers

RTP
  • RTP timeout:
Max RTP timeout
(ex. All calls (if not on hold) will be terminated if there is no RTP activity for the number of seconds set here (60 for example))
([0-9])
  • RTP hold timeout:
Max RTP hold timeout.
NOTE: This field must be a higher number than set under 'RTP timeout'
(ex. All calls on hold will be terminated if there is no RTP activity for the number of seconds set here (300 for example))
([0-9])
  • RTP keep-alive:
Send keep-alives in the RTP stream to keep NAT open (default is off - zero).
(ex. 0)
([0-9])

SIP Timers

SIP Session Timers
  • Default T1 Timer:
Default T1 timer.
Defaults to 500 ms or the measured round-trip time to a peer (qualify=yes).
  • Call Setup Timer
Call setup timer.
If a provisional response is not received in this amount of time, the call will autocongest. Defaults to 32000 ms.

SIP Session Timers

  • Session Timers mode:
Session-Timers feature operates in the following three modes:
originate: Request and run session-timers always
accept: Run session-timers only when requested by other UA
refuse: Do not run session timers in any case
The default mode of operation is 'accept'.
  • Max session refresh interval:
Maximum session refresh interval in seconds.
Defaults to 1800 secs.
(|0-9|)
  • Minimum session refresh interval:
Minimum session refresh interval in seconds.
Defaults to 90 secs.
(|0-9|)
  • Session refresher:
The session refresher (uac|uas).
Defaults to 'uas'.
uac - Default to the caller initially refreshing when possible.
uas - Default to the callee initially refreshing when possible.

DTMF

DTMF
  • DTMF Mode:
Choices are inband, rfc2833, info or auto:
- inband: The device that you press the key on will generate the DTMF tones. If the codec is not ulaw or alaw then the DTMF tones will be distorted by the audio compression and will not be recognised. If the phone is set for RFC2833 and asterisk is set for inband then you may not hear anything.
- rfc2833: http://www.ietf.org/rfc/rfc2833.txt
- info: See SIP method info and SIP info DTMF or http://www.ietf.org/rfc/rfc2976.txt
- auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones if the remote side does not indicate support of rfc2833 in SDP. This feature was added on Sep 6, 2005 and is not available in Asterisk 1.0.x.
NOTE:
Inband DTMF won't work unless the codec is ulaw or alaw (G711). Use out of band DTMF aka rfc2833 or info.
(Select box)
  • Relax DTMF:
Relax DTMF handling
(ex. Set this field to 'Yes' if having problems with DTMF modes)
(Option buttons)

Quality of Service

Quality of Service
  • TOS SIP:
(Select Box)
  • TOS Audio:
(Select Box)
  • TOS Video:
(Select Box)
Available options: CS0, CS1, CS2, CS3, CS4, CS5, CS6, CS7, AF11, AF12, AF13, AF21, AF22, AF23, AF31, AF32, AF33, AF41, AF42, AF43, EF
  • 802.1p CoS SIP:
(Select Box)
  • 802.1p CoS Audio:
(Select Box)
  • 802.1p CoS Video:
(Select Box)
Available Options: 0-7

Misc

Misc
  • Send Diversion Header:
New PBXware version includes new SIP header which will display standard caller ID details along the information on number call is forwarded from.
"From" and "Via" information are enabled by default but can be disabled by setting this option to No. (Option: Yes, No).
  • Max-Forwards value:
Max-Forwards header limits the number of hops a request can make on the way to its destination. It consists of an integer that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request reaches its destination, it is rejected with a 483 (Too Many Hops) error response.
Default value is 70
(|0-9|)
  • Generate inband ringing:
Inband means that DTMF is transmitted within the audio of the phone conversation, i.e. it is audible to the conversation partners. Set whether the system generates in-band ringing.
(ex. You're recommended to set this option to 'Never')
(Select box)
  • Compact Headers:
You can set compact headers to yes or no. If it's set to yes, the SIP headers will use a compact format, which may be required if the size of the SIP header is larger than the maximum transmission unit (MTU) of your IP headers, causing the IP packet to be fragmented. Do not use this option unless you know what you are doing.
(ex. You're recommended to set this option to 'No' unless required otherwise)
(Option buttons)

Authentication

Authentication
  • User Agent:
Set the 'User Agent' string
(ex. 'Custom string')
([a-z][0-9])
  • Realm:
Realm for digest authentication
(ex. 'Custom string')
([a-z][0-9])

Registration

Registration
  • Default Expiration:
Default duration (in seconds) for incoming/outgoing registrations.
Default value 3600 seconds.
([0-9])
  • Minimum Expiration:
Minimum duration (in seconds) for incoming registrations
Default value 60 seconds.
([0-9])
  • Maximum Expiration:
Maximum duration (in seconds) for incoming registrations.
Default value 7200 seconds.
([0-9])
  • Max Contacts
Maximum number of contacts per extension. This number is set as default globally.
This can be overwritten by changing max contacts number per Extension.
([0-9])

MWI

MWI
  • MWI From: header:
When sending MWI NOTIFY requests, use this setting in the From: header as the "name" portion. Also fill the "user" portion of the URI in the From: header with this value if no fromuser is set. Default: empty
([a-z][0-9])
  • Outgoing MWI Expiry:
Expiry time for outgoing MWI subscriptions (seconds).
Default value 3600.
(|0-9|)

Subscriptions

Subscriptions
  • Allow Subscriptions:
Disable support for subscriptions. Default is Yes.
(ex. Yes, No, N/A)
(Option buttons)

Codecs

Codecs
  • Auto-Framing (RTP Packetization):
If auto framing is turned on, the system will choose packetization level based on remote ends preferences.
(ex. Yes, No, N/A)
(Option buttons)

Session Description Protocol

Session Description Protocol
  • T.38 support
Setting this field to any value will enable T.38 FAX support on SIP calls with different options. Default value is off.
  • Session name
This field allows you to change the SDP session name string, Like the useragent parameter, the default user agent string also contains the Asterisk version.
([A-Z, 0-9])
  • Session owner
This field allows you to change the username field in the SDP owner string, This field MUST NOT contain spaces.
([A-Z, 0-9])

Additional Config

This option is used for providing additional config parameters for SIP configuration files. Values provided here will be written into these configuration files.

Additional Config





IAX

IAX (Inter asterisk exchange) is a simple, low overhead and low bandwidth VoIP protocol designed to allow multiple PBXwares to communicate with one another without the overhead of more complex protocols.

General

IAX UAD
  • Port:
IAX bind port
(ex. 4569, (default))
([0-9])
  • Bind Address:
This allows you to bind IAX to a specific local IP address instead of binding to all addresses. This could be used to enhance security if, for example, you only wanted IAX to be available to users on your LAN.
(ex. 0.0.0.0, (default))
([0-9])
  • IAX compatible:
Should layered switches or some other scenario be used
(ex. Set to yes if you plan to use layered switches or some other scenario which may cause a delay when doing a lookup in the dial plan)
(Select box)
  • Language:
Default language settings for all users/peers
(ex. Set this option to 'en' (English), for example)
([a-z])
  • Bandwidth:
Set the bandwidth to control which codecs are used in general
(ex. Select between low, mid, or high)
(Select box)
  • Maximum number of IAX helper threads:
Establishes the number of extra dynamic threads that may be spawned to handle I/O.
(ex. 150)
([0-9])
  • Allow IAXy firmware download:
Controls whether this host will serve out firmware to IAX clients which request it.
(ex. Yes, No, N/A)
(Option buttons)

JitterBuffer

Jitter Buffer
  • Jitter Buffer:
Turn off the jitter buffer for this peer
(ex. Yes, No, N/A)
(Option buttons)
  • Force Jitter Buffer:
Should we force jitter buffer (default value 10)
(ex. Jitter buffer is usually handled by the UADs/Phones. But in case, if these do this poorly, the jitter buffer can be enforced on PBXware side)
([0-9])
  • Max. jitterbuffer interpolations:
The maximum number of interpolation frames the jitter buffer should return in a row
(ex. 1000)
([0-9])
  • Max. Jitter buffer:
A maximum size for the jitter buffer. Setting a reasonable maximum here will prevent the call delay from rising to silly values in extreme situations; you'll hear SOMETHING, even though it will be jittery.
(ex. 1000)
([0-9])
  • Resync Treshold:
Resync the threshold for noticing a change in delay measured
(ex. 1000)
([0-9])

Call Rating

Call Rating
  • AMA Flags:
AMA = Automated Message Accounting.+
These flags are used in the generation of call detail records (e.g 'default')
- default: Sets the system default.
- omit: Do not record calls.
- call rating: Mark the entry for Call Rating
- documentation: Mark the entry for documentation
(Select box)
  • Account code:
Default account for CDRs (Call Detail Records)
(ex. lars101)
([a-z][0-9])

Authorization

Authorization
  • Auth debugging:
Should authentication be debugged
(ex. Setting this option to 'Yes' will increase the amount of debugging traffic)
(Option buttons)
  • Max Auth requests:
Maximum number of outstanding authentication requests waiting for replies. Any further authentication attempts will be blocked
(ex. 10)
([0-9])
  • Delay Reject:
Set this option to 'Yes' for increased security against brute force password attacks
(ex. Yes)
([0-9])

Registration

Registration
  • Registration context:
If the specified PBXware will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us
(ex. iaxregistration)
([a-z][0-9])
  • Min Registration Expire:
Minimum amounts of time that IAX peers can request as a registration expiration interval (in seconds).
(ex. 60)
([0-9])
  • Max Registration Expire:
Maximum amounts of time that IAX peers can request as a registration expiration interval (in seconds).
(ex. 60)
([0-9])

Trunk

Trunk
  • Trunk frequency:
How frequently to send trunk msgs (in ms)
(ex. 20)
([0-9])
  • Trunk Timestamps:
Should we send timestamps for the individual sub-frames within trunk frames
(ex. Yes)
(Option buttons)

Misc

Misc
  • Disable UDP checksums:
Should checksums be calculated
(ex. Yes)
(Option buttons)
  • Auto-kill:
If no response is received within 2000ms, and this option is set to yes, cancel the whole thing
(ex. Yes)
(Option buttons)

Codecs

Codecs
  • Codec Priority:
This option controls the codec negotiation of an inbound IAX calls.
Example:
  • caller - Consider the callers preferred order ahead of the host's.
  • host - Consider the host's preferred order ahead of the caller's.
  • disabled - Disable the consideration of codec preference altogether (this is the original behaviour before preferences were added)
  • reqonly - Same as disabled, but does not consider capabilities if the requested format is not available the call will only be accepted if the requested form
(Read only)

Additional Config

Additional Config

This option is used for providing additional config parameters for IAX configuration files.

Values provided here will be written into these configuration files.

Providers

PBXware comes with a range of pre-configured VoIP and PSTN service providers in order to allow an easy way of adding trunks into the system. This screen allows for the addition of custom providers by clicking on 'Add Custom Provider'.

NOTE: PBXware 5.1 does not support Woomera / BRI any more.

In addition, 'Import Providers' allows for an update of currently pre-configured service providers.

Providers
  • Provider:
Provider name
(ex. Generic Analog)
(Display)
  • Protocol:
Protocol provider uses
(ex. DAHDI)
(Display)
  • Type:
Service type
(ex. pstn/voip)
(Display)
  • edit.png Edit the Provider configuration
(ex. Click to edit the Provider configuration)
(Button)
  • delete.png Delete Provider configuration
(ex. Click to delete Provider configuration from the system)
(Button)

PSTN

DAHDI

DAHDI
  • Type:
Service type
(ex. pstn/voip)
(Select box)
  • Protocol:
Protocol provider uses
(ex. DAHDI)
(Select box)
  • Name:
Provider name
(ex. BT)
([a-z][0-9])
  • Country:
Destination of the trunk connection
(ex. USA)
(Select box)
  • National Dialing Code:
National dialing code at the Provider destination
(ex. For USA 1, United Kingdom, Germany 0)
(0-9)
  • International Dialing Code:
International dialing code at the Provider destination
(ex. For USA 011, United Kingdom, Germany 00)
([0-9])
  • E164 Accepted:
Does the Provider support dialing destinations in the E164 format
(ex. Enabling this option will reformat any dialed number into the following form COUNTRY_CODE+AREA_CODE+DIALED_NUMBER. For example, if you dial 55510205, system will dial 121255510205)
(Option buttons)
  • Pass-thru mode:
If this option is enabled, the number which is dialed is passed through trunk without modification
(ex. Yes, No, N/A)
(Option buttons)
  • Leave National Code:
In some countries, the national code is stripped automatically. If set to 'Yes', the national code will not be stripped from the dialed number.
NOTE: Before setting this option to 'Yes', go to 'Settings: Tenants' and enable this option as well).
(ex. John dials 121255510205. With this option enabled)
([0-9])
  • Local Area Code:
Add the local area code to the dialed number if required by your service provider. (By default, the local area code is stripped when dialing)
(ex. The user dials 55510205, and the local area code is 212. If the call goes through this trunk, PBXware will dial 21210205)
([0-9])
  • Write dialing code:
Should the National and International prefix be written into configuration files
(ex. Enable this option if required by the provider)
([0-9])
  • DAHDI Devices:
Select DAHDI device system is to use
Example:
  • X100P
  • TDM400P
  • TDM10B
  • TDM20B
  • TDM30B
  • TDM40B
  • TDM11B
  • TDM12B
  • TDM13B
  • TDM21B
  • TDM22B
  • TDM23B
  • T100P
  • E100P
  • TE110P
  • TE410P
  • TE405P
  • hfcISDN
(Option buttons)


TIP

Please configure DAHDI devices after you save the provider by clicking the 'Configure' button next to a selected device.

VoIP

This is where you select which type of VoIP you are going to a create a custom provider for.

SIP

SIP

General

General
  • User Type:
User's relationship to the system
Example:
  • user - Provider accepts incoming calls only
  • peer - Provider makes outgoing calls only
  • friend - Provider does both incoming and outgoing calls
(Select box)
  • DTMF Mode (Dual Tone Multi-Frequency):
DTMF mode used by the provider. A specific frequency (consisting of two separate tones) to each key so that it can easily be identified by a microprocessor
Example:
  • inband - inband audio (requires 64 kbit codec - alaw, ulaw)
  • rfc2833 - default
  • info - SIP INFO messages
(Select box)
  • Country:
Destination of the trunk connection
(ex. USA)
(Select box)
  • National Dialing Code:
National dialing code used at the Provider destination
(ex. For USA 1, United Kingdom and Germany 0)
([0-9])
  • International Dialing Code:
International dialing code used at the provider destination
(ex. For USA 011, United Kingdom and Germany 00)
([0-9])
  • E164 Accepted:
Does the trunk support dialing destinations in E164 format
(ex. Enabling this option will reformat any dialed number into the following form COUNTRY_CODE+AREA_CODE+DIALED_NUMBER. For example, if you dial 55510205, the system will dial 121255510205)
(Option buttons)
  • Pass-thru Mode:
If this option is enabled, the number which is dialed is passed through trunk without modification
(ex. Yes, No, N/A)
(Option buttons)
  • Leave National Code:
In some countries, the national code is stripped automatically. If set to 'Yes', national code will not be stripped from the dialed number.
NOTE: Before settings this option to 'Yes', go to 'Settings: Servers' and enable this option as well.
(ex. John dials 121255510205. With this option enabled)
([0-9])
  • Local Area Code:
Add a local area code to the dialed number if required by your service provider. (By default, local area code is stripped when dialing)
(ex. User dials 55510205, local area code is 212. If the call goes through this provider, PBXware will dial 21210205)
([0-9])

Network Related

Network Related
  • Transport:
Type of transfer protocol that will be used on PBXware.
UDP (User Datagram Protocol) - With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without prior communications to set up special transmission channels or data paths.
TCP (Transmission Control Protocol) - provides reliable, ordered, error-checked delivery of a stream of octets between programs running on computers connected to an intranet or the public Internet.
TLS (Transport Layer Security) - cryptographic protocol that provide communication security over the Internet.[1] They use asymmetric cryptography for authentication of key exchange, symmetric encryption for confidentiality, and message authentication codes for message integrity.
Type: Checkbox
  • Encryption:
This option enables or disables encryption in PBXware transport.
Options: Yes, No, N/A.
  • Direct media:
Should you allow RTP voice traffic to bypass Asterisk.
Options:
  • No - this option tells the Asterisk to never issue a reinvite to the client
  • Yes - send reinvite to the client
  • No NAT only - allow reinvite when local, deny reinvite when NAT
  • Use UPDATE - use UPDATE instead of INVITE
  • No NAT, Update - use UPDATE when local, deny when NAT
NOTE: All enhanced services for the extension have to be disabled
(ex. Some devices do not support this, especially if one of them is behind a NAT)
(Option buttons / Select box)
  • Direct RTP setup:
Here you can enable or disable the new experimental direct RTP setup.
Setting this value to yes sets up the call directly with media peer-2-peer without re-invites. Will not work for video and cases where the callee sends RTP payloads and fmtp headers in the 200 OK that does not match the callers INVITE. This will also fail if directmedia is enabled when the device is actually behind NAT.
Options: Yes, No, N/A
  • Default IP:
IP address to be used until registration
(ex. 192.168.1.1)
(IP Address)

Channels

Channels
  • Incoming Limit:
Number of simultaneous incoming calls that the Provider can handle
(ex. 4 = four simultaneous incoming calls. Any additional calls will get the busy sound)
([0-9])
  • Outgoing Limit:
Number of simultaneous outgoing calls that the Provider can handle
(ex. 4 = four simultaneous outgoing calls. Any additional calls attempting to use this Provider will be rejected or will be redirected to other Providers depending on what is set in the system/extensions)
([0-9])
  • Busy level:
Limits the number of concurrent incoming active calls.
  • Email on exceeded limit
Send an e-mail when the incoming/outgoing limit is reached
(ex. Yes, No, N/A)
(Option buttons)
  • Outgoing Dial Options
Advanced dial options for all outgoing calls
(ex. Please see below for a detailed list of all available dial options( default: empty ))
([a-z])

Authentication

Authentication
  • Host:
Provider IP address
(ex. Enter the host IP, 192.168.1.1 for example, or set 'dynamic' if the host is behind dynamic IP address)
([0-9][a-z])
  • Peer Host:
The IP of a peer host to which the system sends the calls
(ex. 192.168.1.1)
(IP Address)
  • Auth:
Global credentials for outbound calls, i.e. when a proxy challenges your PBXware for authentication these credentials override any credentials in peer/register definition if realm is matched.
(ex. john:dfgERGf@pbxware)
([a-z][0-9])
  • Register:
Method for registering to the remote server
(ex. Providers may require a different method of registration to their server. You may choose between 'registration not required', 'register with phone number', and 'register with username')
(Select box)
  • Register suffix:
Service provider may request different registration methods for their services. Select the proper method, as required by the provider
(ex. 1234567)
([0-9])
  • Insecure:
Which of the selected options are not used for authentication.
(ex. very - Do not require authentication of incoming INVITEs)
(Select box)

Codecs

Codecs


Available Codecs:

  • ITU G.711 ulaw - 64 Kbps, sample-based, used in US
  • ITU G.711 alaw - 64 Kbps, sample-based, used in Europe
  • ITU G.722
  • ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
  • ITU G.726 - 16/24/32/40 Kbps
  • ITU G.726 AAL2
  • ITU G.729 - 8 Kbps, 10ms frame size
  • GSM - 13 Kbps (full rate), 20ms frame size
  • iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
  • Speex - 2.15 to 44.2 Kbps
  • LPC10 - 2.5 Kbps
  • H.261 Video - Used over ISDN lines with resolution of 352x288
  • H.263 Video - Low-bit rate encoding solution for video conferencing
  • H.263+ Video - Extension of H.263 that provides additional features that improve compression over packet switched networks.
  • H.264 Video
  • Video Support:
This option enables or disables PBXware video support.
Options: Yes, No, N/A.
  • Auto-Framing (RTP Packetization)
Automatically adjust the RTP packet size with the registered device.
(ex. Yes)
(Option buttons)

IAX

IAX
  • User type:
User's relationship to the system
Example:
  • user - Provider accepts incoming calls only
  • peer - Provider makes outgoing calls only
  • friend - Provider does both incoming and outgoing calls
(Select box)
  • DTMF Mode (Dual Tone Multi-Frequency):
DTMF mode used by the provider. A specific frequency (consisting of two separate tones) to each key so that it can easily be identified by a microprocessor.
Example:
  • inband - inband audio(requires 64 kbit codec - alaw, ulaw)
  • rfc2833 - default
  • info - SIP INFO messages
(Select box)
  • Country:
Destination of the trunk connection.
(ex. USA)
(Select box)
  • National Dialing Code:
National dialing code used at the Provider destination
(ex. For USA 1, United Kingdom, Germany 0)
([0-9])
  • International Dialing Code:
International dialing code used at the provider destination
(ex. For USA 011, United Kingdom, Germany 00)
([0-9])
  • E164 Accepted:
Does the trunk support dialing destinations in the E164 format
(ex. Enabling this option will reformat any dialed number into the following form COUNTRY_CODE+AREA_CODE+DIALED_NUMBER. For example, if you dial 55510205, the system will dial 121255510205)
(Option buttons)
  • Pass-thru Mode:
If this option is enabled, the number which is dialed is passed through the trunk without modification
(Yes, No, N/A)
(Option buttons)
  • Leave National Code:
In some countries, the national code is stripped automatically. If set to 'Yes', the national code will not be stripped from the dialed number.
NOTE: Before setting this option to 'Yes', go to 'Settings: Servers' and enable this options as well.
(ex. John dials 121255510205. With this option enabled)
([0-9])
  • Local Area Code:
Add a local area code to the dialed number if required by your service provider. (By default, local area code is stripped when dialing)
(ex. User dials 55510205, local area code is 212. If the call goes through this provider, PBXware will dial 21210205)
([0-9])

Network Related

Network Related
  • Default IP:
IP address to be used until registration
(ex. 192.168.1.1)
(IP Address)

Channels

Channels
  • Incoming Limit:
Number of simultaneous incoming calls that the Provider can handle
(ex. 4 equals to four simultaneous incoming calls. Any additional calls will get the busy sound)
([0-9])
  • Outgoing Limit:
Number of simultaneous outgoing calls the Provider can handle
(ex. 4 equals to four simultaneous outgoing calls. Any additional calls attempting to use this Provider will be rejected or will be redirected to other Providers depending on what is set in the system/extensions)
([0-9])
  • E-mail on exceeded limit
Send an e-mail when the outgoing limit is reached
(ex. Yes, No, N/A)
(Option buttons)
  • Outgoing Dial Options
Advanced dial options for all outgoing calls
default: empty
([a-z])
  • Notransfer:
Disable native IAX transfer
(Option buttons)
  • Send ANI:
Should ANI ("super" Caller ID) be sent over this Provider
(ex. Set 'Yes' to enable)
(Option buttons)
  • Trunk:
Use IAX2 trunking with this host
(ex. Set 'Yes' to enable)
(Option buttons)

Authentication

Authentication
  • Encryption:
Should encryption be used when authenticating with the peer
(Select box)
  • Host:
Provider IP address
( ex. Enter host IP, 192.168.1.1 for example or set 'dynamic' if host is behind dynamic IP address)
([0-9][a-z])
  • Peer Host:
IP of a peer host system sends the calls to
(ex. 192.168.1.1)
((IP Address)
  • Register:
Method for registering to the remote server
(ex. Providers may require different way of registration to their server. You may choose between 'registration not required', 'register with phone number', and 'register with username')
(Select box)
  • Register suffix:
Service provider may request different registration methods for their services. Select the proper method as required by the provider
(ex. 1234567)
([0-9])
  • Auth Method:
Authentication method required by provider.
The least secure is "plaintext", which sends passwords cleartext across the net.
"md5" uses a challenge/response md5 sum arrangement, but still requires both ends have plain text access to the secret.
"rsa" allows unidirectional secret knowledge through public/private keys. If "rsa" authentication is used, "inkeys" is a list of acceptable public keys on the local system that can be used to authenticate the remote peer, separated by the ":" character.
"outkey" is a single, private key to use to authenticate to the other side.
([a-z][0-9])
  • RSA key:
RSA authentication key
(ex. If Auth Method is set to RSA, then provide the RSA key here)
([a-z][0-9])

Codecs

Codecs
Available Codecs:
  • ITU G.711 ulaw - 64 Kbps, sample-based, used in US
  • ITU G.711 alaw - 64 Kbps, sample-based, used in Europe
  • ITU G.722
  • ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
  • ITU G.726 - 16/24/32/40 Kbps
  • ITU G.726 AAL2
  • ITU G.729 - 8 Kbps, 10ms frame size
  • GSM - 13 Kbps (full rate), 20ms frame size
  • iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
  • Speex - 2.15 to 44.2 Kbps
  • LPC10 - 2.5 Kbps
  • H.261 Video - Used over ISDN lines with resolution of 352x288
  • H.263 Video - Low-bit rate encoding solution for video conferencing
  • H.263+ Video - Extension of H.263 that provides additional features that improve compression over packet switched networks.
  • H.264 Video

E-mail Templates

Email Templates

Here you can decide how the mails that are sent by the system should look. By default, several macros are used and if you want to modify the template using them, they are pretty self explanatory.


  • New Extension Templates
Set e-mail templates which are used to send basic info on newly created extension.

NOTE: Since PBXware 4, e-mail that is sent to user when Save & E-mail button is pressed, will provide you e-mail and user password, used for gloCOM registration. However, in some cases, when phones are not auto provisioned, users still require deskphone registration details to be sent which was not possible at that moment.

For this reason, we have added variables for SIP User ID - %USERNAME% and extension secret -  %SECRET% which can be used to add these information to the e-mail template if required. Introduced the variable %PROXY% to Email Templates that references the value of gloCOM SIP Proxy in tenant settings.

New Extension Templates








  • Call Rating Templates
This is used to set E-mail template for extensions' Soft and Hard Call Rating limits.
Call Rating E-mail Templates






  • Incoming/Outgoing Limit Templates
Set the e-mail template used for sending emails when an extension reaches its limit for incoming/outgoing number of channels.
Incoming/Outgoing Limit Templates







  • Fax To E-mail Template
Set the e-mail template used for sending Fax To E-mail when fax is received.
Fax to E-mail Template






  • Instant Recording Template
Set the e-mail template used for sending out instant recording e-mails.
Instant Recording Template






  • Voicemail E-mail Template
Set the e-mail template used for sending emails when new Voicemail is received.
Voicemail E-mail Template

E-mail Notifications

E-mail Notifications

Here you can choose which notifications will you receive on email.

  • Extension configuration details:
Check this option if you want to receive configuration details of extensions you create on your system.
  • PBX service check notifications:
Whether you want to receive notifications on PBX services.
  • Filesystem check configuration:
Whether you want to receive filesystem notifications.
  • Channel limits notifications:
Check this option if you want to receive notifications on your channel limits.
  • Call Rating notifications:
Whether you want to receive notifications on call rating.

Voicemail

Calls are diverted to Voicemail when a user is unavailable or has the phone powered off, or when a call is transferred to voicemail by user. The phone alerts the user to indicate the receipt of a message.

Once the user is transferred to the party's voice box a 'Please leave a detail message after the tone. If you would like to speak to the operator, press 0' message will be heard.

The user has two options:

  1. To leave a voice message that is ended by pressing the # key or by hanging up
  2. To reach an operator by dialing 0

If 0 is dialed a 'Press 1 to accept this recording, otherwise please continue to hold' message will be heard. The user has two options:

  1. Press 1 to save your message and dial the operator. 'Please hold while I try that extension' message played.
  2. Continue to hold to delete your message and dial the operator. 'Message deleted, please hold while I try that extension' message played.


General Voicemail

General fields are most required by voicemail

General Voicemail
  • Format:
Audio format in which voice messages are recorded
(ex. If 'wav49' is selected here, all voice messages will be saved in this format. See below for disk usage)
(Select box)
  • Max Message Length:
Maximum length of a voice message in seconds
(ex. By default this field is set to '180' seconds (3 minutes)
([0-9])
  • Min Message Length:
Minimum length of a voice message in seconds
(ex. Default value set to '3' seconds. Messages that last less are discarded)
([0-9])
  • Max Greeting Length:
Maximum length in seconds of the user recorded voicemail greeting message
(ex. Default values set to '60' seconds)
([0-9])
  • Max Seconds of Silence:
Maximum length in seconds of silence in a voice message
(ex. Default value set to '10' seconds. Silence longer than set here will end a voice message)
([0-9])
  • Silence Threshold:
Silence detection threshold
(ex. Default value set to '128'. The higher the number, more background noise is added)
([0-9])
  • Voicemail Delay:
Delay a number of seconds before asking user for 'Password'
(ex. If you hear a partial sound file played asking the user for the password, set '1' or '2' here to add a second or two of silence before the sound file is played).
([0-9])
  • Max Files per Directory:
Maximum number of voicemail messages per voicemail directory
(ex. Each voice box has following directories (INBOX, Old, Work, Family, Friends, Cust1, Cust2, Cust3, Cust4, Cust5). Set this field to '100' to allow a 100 voice messages per each voice directory)
([0-9])
  • Min PIN Length:
Defines the minimum length of PIN used for Voicemail
(ex. 4)
([0-9])

Disk Space Used By Voicemail Recording

With continuous tone for 60 seconds:

  • wav49 = 91.0kb
  • wav = 863.0kb
  • gsm = 91.0kb

With continuous silent tone (without sound) for 60 seconds:

  • wav49 = 0.38kb
  • wav = 3.0kb
  • gsm = 0.32kb

E-mail Settings

Customize the display of emails that notify users of new voicemail messages.

Email Settings
  • Server E-mail:
This email address is used to identify from whom the email came
(ex. If this field is set to 'pbxe@domain.com', the following line is added in the email header '"$FROM": string <pbxe@domain.com>')
([a-z][0-9])
  • Send Attachment:
Send the voice message as an attachment to the user's email.
(ex. Once B gets the new voice message, if this option is set to 'Yes', the message sound file will be attached to the new voicemail notification email).
(Option buttons)
  • Delete After E-mailing:
Delete the voice message after sending it as an attachment to the user's email.
(ex. Once B gets the new voice message, if this option is set to 'Yes', the message will be deleted from the voice box after it has been emailed to B).
(Option buttons)
  • Skip[PBX]: in subject:
Should 'PBX' be skipped from voicemail title
(ex. If set to 'No' - 'Subject: [PBX']: New message M in mailbox B' will be displayed in the email subject line)
(Option buttons)
  • Charset:
Charset that is in use on PBXware.
To display an HTML page correctly, a web browser must know what character set to use.
(default value 'ISO-8859-1')

Run Application

Run custom applications on certain voicemail actions

Run Application
  • On Voicemail:
Run a custom application when a new voicemail is received
(ex. Set this field to '/usr/bin/myapp' to execute 'myapp' application when a new voicemail arrives)
([a-z][0-9])
  • On Password Change:
Run a custom application when your voicemail password is changed
(ex. Set this field to '/usr/bin/myapp' to execute 'myapp' application when your voicemail password changes)
([a-z][0-9])

Main Voicemail Menu

Edit settings for main voicemail menu.

Main Voicemail Menu
  • Play Envelope message:
Announces the Date/Time and the Extension number from which the message was recorded.
(ex. Once voice box is checked for new messages, if this option is set to 'Yes', 'Received at {$DATE}. The from phone number {$NUMBER}' will be played, giving more details about the message originator).
(Option buttons)
  • Say Caller ID:
Announce the extension number from which the voice message was recorded.
(ex. If this option is set to 'Yes', when checking voicemail, 'From phone number {$NUMBER}' message will be heard).
(Option buttons)
  • Skip ms on playback:
Interval in milliseconds to use when skipping forward or reverse while a voicemail message is being played
(ex. If this field is set to '3000', when listening to a voice message it will skip 3 seconds on rewind/fast forward)
([0-9])
  • Max login attempts:
Maximum number of login retries before user gets disconnected
(ex. By default, this field is set to '3'. After 3 unsuccessful login attempts, the user gets disconnected)
([0-9])
  • On Delete, play next message:
After a voice message has been deleted, should the system automatically play the next message from the voice inbox
(ex. Select 'Yes' to automatically playback the next voice message after you've deleted the old one)
(Option buttons)

Advanced Features

Advanced Features
  • Allow Review mode:
Allow B to review the voice message before committing it permanently to A's voice box.
Example:
B leaves a message on A's voice box, but instead of hanging up, he presses '#'. Three options are offered to B:
  • Press 1 to accept this recording
  • Press 2 to listen to it
  • Press 3 to re-record your message
(Option buttons)
  • Allow Operator:
Allow B to reach an operator from within the voice box.
Example:
B leaves a message on A's voice box, but instead of hanging up, B presses '#'.
'Press 0 to reach an operator' message is played (Once '0' is pressed, the user is offered the following options):
  • Press 1 to accept this recording (If selected, 'Your message has been saved. Please hold while I try that extension' is played and operator is dialed)
  • Or continue to hold (If B holds for a moment, 'Message deleted. Please hold while I try that extension' is played and operator is dialed)
(Option buttons)
  • System Operator:
Local extension number that acts as an operator.
(ex. If A's voice box has an option 'Allow Operator' set to 'Yes', all users dialing '#0' inside the voice box will reach this operator extension).
([0-9])
  • Send Voicemail:
Change the context from which to send voicemail
(ex. Select 'Yes' to enable and provide new values to 'Dialout context' and 'Callback context' fields)
(Option buttons)
  • Volume Gain:
Set the gain you want to apply to VoiceMail recordings.
(ex. Enter 2 to amplify the recordings).
([0-9])
  • Dialout context:
Context to dial out from
(ex. Set this field to 'fromvm', for example)
([a-z][0-9])
  • Callback context:
Context to call back
(ex. Set this field to 'tomv'. If not listed, calling the sender back will not be permitted)
([a-z][0-9])
  • Allow Voicemail Menu (via *):
Allow caller to access Voicemail Menu by dialing *.
(ex. When reaching extensions voicemail to leave a message, pressing * will present you with login for extensions Voicemail box.).
  • Allow Voicemail Dialout and Callback:
Allow caller to use Voicemail Dialout and Callback from voicemail menu by entering extensions PIN.
  • Timezone:
Timezone that system will use to inform users when voicemail message was received. Timezone selected here will also be used for setting up timezone for auto provisioned devices.

Edit Timezones

Edit Timezones
  • Timezone:
Timezone name
(ex. Bosnia and Herzegovina)
(Display)
  • edit.png Edits the timezone configuration
(ex. Click to edit timezone configuration)
(Bitton)
  • delete.png Deletes a timezone from the system
(ex. Click to delete a timezone from the system)
(Button)

Add/Edit Timezones

Add/Edit Timezones
  • Name:
Unique timezone name
(ex. The name provided here will be visible when setting the correct voicemail timezone. Type 'Zenica' here for example)
([a-z][0-9])
  • Timezone:
Set the correct timezone
(ex. If you have set 'Name'='Zenica' (a town in Bosnia) select the closest timezone to Zenica here (e.g. 'Europe/Sarajevo'))
(Select box)
  • Time format:
Set the appropriate time format:
Example:
Depending on your selected 'Timezone' you may choose between the following options:
  • 12 Hour clock
  • 12 Hour clock including minute
  • 12 Hour clock AM/PM
  • 12 Hour clock AM/PM, including minute
  • 24 Hour clock
  • 24 Hour clock including minute
  • AM/PM 12 hour syntax
  • Dutch syntax
  • German syntax
  • Greek syntax
  • Italian syntax
  • Norwegian syntax
  • Swedish syntax
(Select box)
  • Date format:
Set the correct date format
Example:
Depending on the selected 'Timezone', you may choose between the following options
  • Month/Day/Year
  • Day of Week/Month/Day/Year
  • Day/Month/Year
  • Day of Week/Day/Month/Year
(Select box)
  • Custom sound:
This file is played before the voicemail arrival time
(ex. Enter sound file name, without the extension (e.g. 'arlington') here)
([a-z][0-9])

ADSI

ADSI
  • ADSI Feature Number
The ADSI feature descriptor number to download to
(ex. 0000000F)
([A-F][0-9])
  • ADSI Security code
The ADSI security lock code
(ex. 9BDBF7AC)
([A-F][0-9])
  • ADSI Application version
The ADSI voicemail application version number
(ex. 1)
([0-9])

Transcription

Under voicemail the "Transcribe by Default" option has been implemented. This option will determine if the transcription is enabled on all extensions by default. (Not Set behaves the same as Yes to mimic the behavior of previous versions).
Transcription


Speech to Text for Voicemails

Introduction

This is a feature in PBXware which allows customers to attach the transcript of voicemails when sending out emails to users while also being able to access this using OSC. Currently there are 2 transcription services supported for this feature:

  • Google Speech
  • IBM Watson


Configuration

This feature will disabled by default ands to be configured under: Settings -> Voicemail -> Advanced Options in the Transcription section. Any fields active in this section are mandatory and can’t be empty. Initially only the Enable button will not be disabled:

Speech to Text





A wide variety of languages are supported:

  • English (US)
  • English (UK)
  • Spanish
  • French
  • Bengali
  • Greek

etc.


Depending on the the selected service additional fields appear that have to be filed with the service specific credentials.

NOTE: In order for Speech-to-Text transcription to work, accounts for IBM Watson and Google Speech are to be set up first. IBM Watson (https://www.ibm.com/cloud/watson-speech-to-text) is free of charge and Google Speech (https://cloud.google.com/speech-to-text/) service comes with a small monthly charge.


IBM Watson:

IBM Watson






Google Speech:

Google Speech





Only the information for the currently selected service will be used, the rest will be ignored. In case service got switched the previously used information will be preserved and can be reused when switching back to that service.

NOTE: In order for Speech-to-Text transcription to work Delete After E-mailing has to be disabled on the extension, otherwise no transcription will be generated.


Service credentials acquisition

IBM Watson

In order to get the credentials for IBM Watson’s speech-to-text service an IBM Cloud account is required (free account works as well but limits the amount of requests that can be done). After logging into your IBM Cloud account select Watson from the menu that opens after clicking on the button in the top left corner:

IBM Watson

























In the page that opens under Watson Services select Browse Services (2). From the offered services select Speech to text (2) then press the Add Service button (3).

5.4 watson credentials.png






On the next page under theService Name enter the project name (1). Afterwards select the resource group (2), if multiple are available, as well as the deployment region (3).

You can also add an optional tag to your project (4).
5.4 watson credentials project.png






After these options are filled in scroll to the bottom of the page, there you will select your pricing plan (1) and press Create (2) to create your project.

Once the project is created go to the credentials page of the project (1).

5.4 MT watson credentials service.png






There will be a set of auto generated credentials to be used or new credentials can be made by pressing the New Credentials button (1).

To use either custom or auto-generated credentials press View credentials (2).
5.4 MT watson credentials view.png






In case new credentials are created the following window will appear where you enter the credentials name (1) and role (2).

Optionally the service credential ID (2) and configuration parameters (4) can be added.
Once the mandatory fields are set press Add (5).
5.4 MT watson credentials add new.png






Once a service credential is open for view, only 2 fields are needed for voicemail transcription: apikey (1) and url (2).

5.4 MT watson apikey url.png







Google Speech

Log into google cloud console with an existing account. From the menu on the left select APIs and Services:

5-1-speech10.png











Either click Library or ENABLE APIS AND SERVICES:

5-1-speech11.png






From here either manually find the Google Cloud Speech API or using the search field and select it. On the Google Cloud Speech API page click the enable button:

5-1-speech12.png







After enabling the api press the MANAGE button that replaced the enable button. On the manage page go to the Credentials section (1) and press Create credentials (2) and select API key (3):

5-1-speech13.png







A popup window will appear containing the generated key press the copy icon (1) to copy the API key needed for this feature. Some restrictions can be placed on individual keys (IP address, ios/android apps..) can be placed on the key by pressing the REStRICT KEY button (2).

5-1-speech14.png






Already created keys can be accessed from the same page in the API keyssection (the site might need to be refresh in order for keys to display). IP restriction is recommended and can be set and can be set by selecting IP addresses (1) in the in the Key restriction section of the API key options. Addresses are added separately by typing them in the IP address input field (2) and pressing enter confirm each one. Already added addresses will display above the input field (3) where they can be edited or removed by deleting the rows content or pressing the appropriate x button (4).


5-1-speech15.png







Usedge

Transcript can be added in Voicemail E-mail Template as shown below.

5-1-speech16.png






In order to make this feature easier to be used the default HTML template got updated, adding an additional row with the transcript. That row will be omitted if no transcript is available. An additional option has been added to all e-mail templates that enables the template to be reset to the default one. This reset will affect all the templates for the resource it is used on (eg: if voicemail e-mail templates get reset it will affect the plain text template, html template, and pager template). A warning message will pop up where confirmation for the reset is required in order to evade accidental resets.

5-1-speech17.png







To see the transcript in OSC click the speech bubble left of the message number:

5-1-speech18.png







In case no transcript is available (like voicemails created without this option) that field will be empty. The transcript will appear in a popup above the speech bubble and disappear after the first click outside the popup.

CNAM Lookup

CNAM Lookup

CNAM lookup provides the option to search and display caller name based on CallerID.

The following options are available and required:

  • Enable - must be set to yes in order to use the service when a call comes to the system.
  • Lookup URL - a valid URL given by a CNAM provider; the phone number in the URL must be replaced with “%NUMBER%”.
  • Request method - preferred method for getting a response from a CNAM provider, defined by the provider itself (“get” or “post”).
  • Request timeout (sec) - time limit in seconds for getting response from a CNAM provider
  • Lookup field - a CNAM database field given by a provider that holds the caller data we want to display (example “name”).
  • Test phone number - Any familiar number to be used for testing purposes.

A test request will be performed on every “save” and “test” action and the response will be displayed at the bottom of the CNAM settings form.

Possible CNAM lookup providers to be used:

http://www.cidname.com/

https://www.data24-7.com/

https://www.opencnam.com/


Examples of CNAM lookup links for some providers:

data24-7

https://api.data24-7.com/v/2.0?user=YourUserName&pass=YourPassword&api=I&p1=%NUMBER%&out=json

OpenCNAM

https://api.opencnam.com/v3/phone/+14153545628?account_sid=SID&auth_token=TOKEN&format=json

CID(name)

https://dip.cidname.com/%NUMBER%/46f65a6fd221ee5e3421a666db5dd24a22d17b255e5d1&&output=json&reply=0

Please check with your provider to make sure that CNAM lookup link is in correct format.

Configuration Files

System configuration files are accessible through this section. Only trained users should modify configuration files.

Configuration Files

First, select a configuration file from the right hand side navigation (e.g. FEATURES). The file will open in a big text box from where it can be modified. Once the file is changed, click on the 'Save' button.

If there is a message saying 'Configuration files not updated' visible, click on the 'Import current file' or 'Import All Files' button to update the files. This error usually happens if the configuration file is changed through the system console (shell).


TIP

Once any kind of file update is done, be sure to Reload PBXware to apply the changes.


  • Extensions
BLF for Conference rooms (v5.3.5 update)
e.g.: An assistant rolls calls into conference rooms and when the buttons light up on the executive phones, the executive would know the call is now in the conference room.
The config needs to be set as follows (example for Tenant 200, Conference 119):

Multi Tenant:

[t-200]
exten => 119,hint,confbridge:200119

About

About section displays systems edition, version, release date and licensing information.

About
  • PBXware MT:
This line identifies the PBXware MT version, release date (Revision) and Asterisk running
(ex. Edition: Business, Version: 3.0, Running: 1.4.24-gc-86b7f08)
(Display)
  • Package:
Package information displays a number of system Extensions, Trunks, Conferences ...
(ex. Servers: 1, Extensions 768...)
(Display)
  • Enhanced Services:
This line identifies system Enhanced Services
(ex. Last Caller, Group Hunt, Call Forwarding...)
(Display)
  • License Details:
This line displays license details. NOTE: If 'Branding' is enabled, only license number is visible
(ex. License No: 7E5CF50C)
(Display)


Settings: On slave tenants

Settings: On slave tenants

Settings on a slave are used to set current slaves variables. These settings are applied only on the current slave tenant.






Default Trunks

Slaves use trunks, assigned to them on the master tenant, to place calls to various destinations. In order to allow an organization to control its voice communications budget and to provide for termination backup the default trunks allows setting primary, secondary and tertiary trunks.

The slave tenant will use the primary trunk as its first choice for every destination called. If the primary trunk for some reason fails to terminate the call, the secondary trunk will be used by the system. If the secondary trunk for some reason fails to terminate the call, the tertiary trunk will be used by the system.

Default trunks
  • Primary/Secondary/Tertiary Trunk:
Select default trunks on the system level.
NOTE: If the dialed number is busy, the system will recognize it and won't skip to other trunks in order to dial it.
(ex. Service provider name)
(Select box)
  • Default Destination:
Default destination where trunks without DID will be transferred
(ex. If there is no DID for an incoming Trunk/Provider, all calls will be transferred to this system number (e.g. 2000))
([0-9])


Precedence

Settings:

  • Default Trunks: All System calls go through trunks defined here
  • MiniLCR: Overrides 'Default Trunks' and sets a specific trunk for a destination

Extensions:

  • Trunks: Overrides 'Settings: Default Trunks'
  • Routes: Overrides 'Settings: MiniLCR'

UAD

UADs (User Agent Devices) are various IP phones, soft phones, ATA (Analog Telephone Adaptors), and IAD (Integrated Access Devices) used for system extensions. PBXware supports a wide range of UAD using SIP, IAX, MGCP, and DAHDI protocols.

Supported devices are already pre-configured with the most common settings in order to allow administrators an easy way of adding extensions. However, some PBXware installations have specific requirements, hence it is advisable to edit the selected UAD and set it to the required values. Additionally, if an installation needs to use an unlisted UAD, clicking on "Add User Agent" allows adding new UAD.

UAD


End of Life

All end-of-life devices are now properly marked on the Bicom Systems Wikimedia site under the UAD section.

As for the PBXware GUI:
To view which devices are end-of-life, freely navigate to the Settings tab > select the UAD page and a selection box that says 'End of Life' will be shown. Simply choose 'Yes' if you wish to see the list of end-of-life devices.


End of Life Devices



Requirements

Paging

Paging is a service that supports transmitting of messages to multiple phones over their loudspeakers

NOTE: With addition of multiple registration to a single extension, paging had to be modified. In PBXware 5 if more than one deskphone or third party softphone application is registered to a single PBXware extension at the same time, Speakerphone paging will be disabled for that extension. This is expected behavior and a tradeoff users must accept when registering more than one deskphone to their extension. Registration of Bicom Systems gloCOM applications, desktop and mobile, will not affect paging.


Budgetone 101/102

1. Navigate Internet browser to the phone IP address

2. Provide admin password ('admin' by default)

3. Click on the 'Login' button

4. Click on the 'Advanced Settings' link

5. Scroll down to the 'Auto-Answer' options

6. Select 'Yes'

7. Click on the 'Update' button

8. Click on the 'Reboot' button

TIP

Paging tested with Firmware version 1.0.8.16

GXP 2000

1. Navigate in your Internet browser to the phone IP address

2. Provide the admin password ('admin' by default)

3. Click on the 'Login' button

4. Click on the 'Account *' you wish to edit

5. Scroll down to 'Auto-Answer' options

6. Select 'Yes'

7. Click on the 'Update' button

8. Click on the 'Reboot' button

TIP

Paging tested with Firmware version 1.0.1.12

Snom 190/320

1. Navigate in your Internet browser to the telephone IP address

2. Navigate to your line e.g. 'Line 1'

3. Click 'SIP'

4. Set 'Auto Answer' to 'On'

5. Click on the 'Save' button

6. Navigate to 'Preferences'

7. Set 'Auto Answer Indication' to 'On' for a sound to be played notifying you that a call has been received

8. Set 'Type of Answering' to suit your needs e.g. 'Handsfree'

9. Click on the 'Save' button

TIP

Paging tested with Firmware version 5.2b

Polycom 30x/50x/60x

You need the latest version of both the SIP software and bootROM to do it. Auto-answer could be configured only using provisioning. To prepare configuration files you have to do following steps:

1. In the 'sip.cfg' file, look for the line with these variables:

<alertInfo voIpProt.SIP.alertinfo.1.value="Auto Answer" voIpProt.SIP.alertInfo.1.class="3"...> 

Polycom calls up class 3 in sip.cfg or ipmid.cfg file.

2. In 'sip.cfg' my ring class 'AUTO_ANSWER' looks like this:

<ringType se.rt.enabled="1" se.rt.modification.enabled="1">
<DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="2" se.rt.1.callWait="6" se.rt.1.mod="1"/>
<VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual"/>
<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/>
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>
<INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="2" se.rt.5.callWait="6" se.rt.5.mod="1"/>
<EXTERNAL se.rt.6.name="External" se.rt.6.type="ring" se.rt.6.ringer="2" se.rt.6.callWait="6" se.rt.6.mod="1"/>
<EMERGENCY se.rt.7.name="Emergency" se.rt.7.type="ring" se.rt.7.ringer="2" se.rt.7.callWait="6" se.rt.7.mod="1"/>
<CUSTOM_1 se.rt.8.name="Custom 1" se.rt.8.type="ring" se.rt.8.ringer="5" se.rt.8.callWait="7" se.rt.8.mod="1"/>
<CUSTOM_2 se.rt.9.name="Custom 2" se.rt.9.type="ring" se.rt.9.ringer="7" se.rt.9.callWait="7" se.rt.9.mod="1"/>
<CUSTOM_3 se.rt.10.name="Custom 3" se.rt.10.type="ring" se.rt.10.ringer="9" se.rt.10.callWait="7" se.rt.10.mod="1"/>
<CUSTOM_4 se.rt.11.name="Custom 4" se.rt.11.type="ring" se.rt.11.ringer="11" se.rt.11.callWait="7" se.rt.11.mod="1"/>
</ringType> 

'se.rt.3.type="ANSWER"' sets the Polycom phone ring type, in this case, an answer, that means that the phone will automatically answer without ringing.

3. Update modified files to the provisioning server

4. Reload PBXware if used as the provisioning server

5. Restart your telephone

The bootROM on the phone performs the provisioning functions of downloading the bootROM, the <Ethernet address>.cfg file, and the SIP application and uploading log files. The SIP application performs the provisioning functions of downloading all other configuration files, uploading and downloading the configuration override file and user directory, downloading the dictionary and uploading log files.

The protocol which will be used to transfer files from the boot server depends on several factors including the phone model and whether the bootROM or SIP application stage of provisioning is in progress. TFTP and FTP are supported by all SoundPoint and SoundStation phones. The SoundPoint IP 301, 501, 600, and 601 and SoundStation IP 4000 bootROM also supports HTTP while the SIP application supports HTTP1 and HTTPS. If an unsupported protocol is specified, this may result in unexpected behavior, see the table for details of which protocol the phone will use. The "Specified Protocol" listed in the table can be selected in the Server Type field or the Server Address can include a transfer protocol, for example http://usr:pwd@server (see 2.2.1.3.3 Server Menu on page 10). The boot server address can also be obtained via DHCP. Configuration file names in the <Ethernet address>.cfg file can include a transfer protocol, for example https://usr:pwd@server/dir/file.cfg. If a user name and password are specified as part of the server address or file name, they will be used only if the server supports them.

Polycom

TIP A URL should contain forward slashes instead of back slashes and should not contain spaces. Escape characters are not supported. If a user name and password are not specified, the Server User and Server Password will be used.

For downloading the bootROM and application images to the phone, the secure HTTPS protocol is not available. To guarantee software integrity, the bootROM will only download signed bootROM or application images. For HTTPS, widely recognized certificate authorities are trusted by the phone and custom certificates can be added. See 6.1 Trusted Certificate Authority List on page 151. Using HTTPS requires that SNTP be functional. Provisioning of configuration files is done by the application instead of the bootROM and this transfer can use a secure protocol.

Cisco 7940-7960

1. Create a new line on a Cisco phone and put the configuration into sip.conf as you normally would (go into 'Settings: Call Preferences: Auto Answer (intercom)' and then make the line you've just created as 'auto-answer'.)

2. Here are the contents of /var/lib/asterisk/agi-bin/callall:

#!/bin/sh
cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing

3. Make sure to make the script executable. And then for every extension you have as an auto-answer, have a file like this in /var/lib/asterisk/agi-bin:

Channel: SIP/2006
Context: add-to-conference
WaitTime: 2
Extension: start
Priority: 1
CallerID: Office Pager <5555> 

So, for example, if you have three lines that are configured for automatic answering - SIP/2006, SIP/2007, SIP/2008, you should have three files named 2006-conf, 2007-conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied into the outgoing call spool directory every time you call extension 5555.

4. Now, dial 5555 from any phone and you should have one-way paging.

People who use the pager may have to get used to waiting 1-2 seconds before speaking to allow all the phones to catch up with the audio stream. All of the phones hang up after 20 seconds, regardless of if the person originating the page has stopped talking. Change the AbsoluteTimeout values to increase this interval.

If you want a really confusing loud mess, then change the "dmq" options to "dq" and you'll get an N-way conversation going with everyone who has a phone. Bad.

If you want a really interesting office surveillance tool, change the "dmq" to "dt" and you'll suddenly be listening to all of the extensions in the office, like some kind of mega-snoop tool. Useful for after-hours listening throughout the entire office.


TIP

Paging tested with Firmware version 6.1

Linksys 941

1. Navigate in your Internet browser to phone IP address

2. Select 'Admin Login'

3. Select 'Advanced'

4. Navigate to 'User' tab

5. Set 'Auto Answer Page' to 'Yes'

6. Set 'Send Audio To Speaker' to 'Yes'

7. Click 'Submit All Changes' button

TIP

At the time paging option is set for all lines and works once handset is picked up. Paging tested with Firmware version 4.1.12(a)

Aastra 480i/9133i/9112i

1. Navigate your Internet browser to your telephone IP address

2. Select 'Preferences' under the 'Basic Settings' tab

3. Set 'Microphone Mute' to 'No'

4. Set 'Auto-Answer' to 'Yes'

5. Click on the 'Save Settings' button

6. Reboot the phone to apply the new settings

TIP

At the time, the paging option is set for all lines and works once the handset is picked up. Paging tested with Firmware version 1.3.1.1095

Add/Edit

SIP

Click 'Add User Agent' to add a device or click on the 'Edit' icon next to one, to change its settings.

UAD SIP

General

  • Protocol:
Select protocol UAD(User Agent Device) uses
(ex. If your UAD/Phone uses SIP protocol, select 'SIP' here)
(Select box)
  • Device Name:
Unique device name
(ex. AASTRA 480i
([a-z][0-9])
  • DTMF Mode (Dual Tone Multi-Frequency):
A specific frequency consisting of two separate tones. Each key has a specific tone assigned to it so it can be easily identified by a microprocessor.
(ex. This is a sound heard when dialing digits on touch-tone phones. Each phone has different 'DTMF Mode'. By default, this field is populated automatically for supported devices. If adding other UAD/Phone select between 'inband', , 'rfc2833' or 'info' options
(Option buttons)
  • Context:
Every system extension belongs to a certain system context. The context may be described as a collection/group of extensions. Default context used by the PBXware is 'default' and must not be used by custom extensions.
(ex. default)
([a-z][0-9])
  • Status:
Extension status/presence on the network
If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions)
(Select box)
  • Internal UAD name for Auto-Provisioning
This is where you define how the system will recognize the UAD internally
(ex. aa57i)
(Select box)

Network Related

Network Related
  • Transport:
Type of transfer protocol that will be used on PBXware.
UDP (User Datagram Protocol) - With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without prior communications to set up special transmission channels or data paths.
TCP (Transmission Control Protocol) - provides reliable, ordered, error-checked delivery of a stream of octets between programs running on computers connected to an intranet or the public Internet.
TLS (Transport Layer Security) - cryptographic protocol that provide communication security over the Internet.[1] They use asymmetric cryptography for authentication of key exchange, symmetric encryption for confidentiality, and message authentication codes for message integrity.
Type: Checkbox
  • Encryption:
This option enables or disables encryption in PBXware transport.
Options: Yes, No, N/A.
  • NAT (Network Address Translation):
Set the appropriate Extension - PBXware NAT relation
Example:
If Extension 1000 is trying to register with the PBXware from a remote location/network and that network is behind NAT, select the appropriate NAT settings here:
  • Yes - Always ignore info and assume NAT
  • No - Use NAT mode only according to RFC3581
  • Default(rport) - force_rport - forces RFC 3581 behavior and disables symmetric RTP support
  • Comedia RTP - enables RFC 3581 behavior if the remote side requests it and enables symmetric RTP support
(Option button)
  • Direct media
This option tells the Asterisk server to never issue a reinvite to the client, if it is set to No.
(ex. Select Yes if you want Asterisk to send reinvite to the client)
(Option button)
  • Direct RTP setup:
Here you can enable or disable the new experimental direct RTP setup. Setting this value to yes sets up the call directly with media peer-2-peer without re-invites. Will not work for video and cases where the callee sends RTP payloads and fmtp headers in the 200 OK that does not match the callers INVITE. This will also fail if directmedia is enabled when the device is actually behind NAT.
Options: Yes, No, N/A
  • Qualify
Timing interval in milliseconds at which a 'ping' is sent to the UAD/Phone or trunk, in order to find out its status(online/offline).
In PBXware 5.1 'Qualify' is set to 8000 by default.
(ex. Set this option to '2500' to send a ping signal every 2.5 seconds to the UAD/Phone or trunk. Navigate to 'Monitor: Extensions' or 'Monitor: Trunks' and check the 'Status' field)
([0-9])

Call Properties

Call Properties
  • Ringtime:
UAD/Phone ring time
(ex. Time in seconds that the UAD/Phone will ring before the call is considered unanswered)
([0-9])
  • Incoming Dial Options:
Advanced dial options for all incoming calls
(ex. Please see below for a detailed list of all available dial options (default: tr))
([a-z])
  • Outgoing Dial Options:
Advanced dial options for all outgoing calls
(ex. Please see below for a detailed list of all available dial options( default: empty ))
([a-z])

Channels

Channels
  • Incoming Limit:
Maximum number of incoming calls
(ex. 2)
([0-9])
  • Outgoing Limit:
Maxmimum number of outgoing calls
(ex. 2)
([0-9])
  • Busy Level
Maximum number of concurent incoming calls until user/peer is busy.

Codecs

Codecs
Set the codecs extension is allowed to use.
  • Video Support
Set this option to Yes to enable SIP video support.
(Yes, No, N/A)
  • Auto-Framing (RTP Packetization):
If autoframing is turned on, system will choose packetization level based on remote ends preferences.
(ex. Yes)
(Option buttons)

Auto Provisioning

Auto Provisioning
  • Auto Provisioning:
Enable auto provisioning service for this particular model
(ex. Connect the UAD/Phone to PBXware without any hassle by providing the UAD/Phone MAC address (and optionally adding the Static UAD/Phone IP address and network details))
(Option buttons)
  • DHCP ( Dynamic Hosts Configuration Protocol):
Set whether the UAD/Phone is on a DHCP or a Static IP address
(ex. Set DHCP = Yes if UAD/Phone is on a dynamic or DHCP = No if UAD/Phone is on a static IP address. If on a static IP, you will have to provide more network details in the fields below).
(Option buttons)

Presence

Presence
  • Presence
Presence enables UAD to subscribe to hints which are used for BLF.
(ex. Yes, No, N/A)
(Option buttons)
  • User Agent Auto Provisioning Template:
This option is used for providing additional config parameters for SIP, IAX, and MGCP configuration files. The values provided here will be written into these configuration files.
([a-z][0-9])

IAX

Click 'Add User Agent' to add a device or click the 'Edit' icon next to one to change its settings.

IAX
  • Protocol:
Select protocol UAD (User Agent Device) uses
(ex. If UAD/Phone uses IAX protocol, select 'IAX' here)
(Select box)

General

  • Device Name:
Unique device name
(ex. AASTRA 480i)
([a-z][0-9])
  • DTMF Mode (Dual Tone Multi-Frequency):
A specific frequency, consisting of two separate tones. Each key has a specific tone assigned to it so it can be easily identified by a microprocessor.
(ex. This is a sound heard when dialing digits on touch-tone phones. Each phone has a different 'DTMF Mode'. By default, this field is populated automatically for supported devices. If adding another UAD/Phone select between 'inband', , 'rfc2833' or 'info' options)
(Select box)
  • Context:
Every system extension belongs to a certain system context. Context may be described as a collection/group of extensions. The default context used by PBXware is 'default' and must not be used by custom extensions.
(ex. default)
([a-z][0-9])
  • Status:
Extension status/presence on the network
(ex. If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions)
(Select box)

Network Related

Network Related
  • NAT (Network Address Translation)
Set the appropriate Extension - PBXware NAT relation.
If extension 1000 is trying to register with the PBXware from a remote location/network and that network is behind NAT, select the appropriate NAT settings here:
  • yes - Always ignore info and assume NAT
  • no - Use NAT mode only according to RFC3581
  • Default (rport) - this setting forces RFC3581 behavior and disables symmetric RTP support.
  • Comedia RTP - enables RFC3581 behavior if the remote side requests it and enables symmetric RTP support.
(Option buttons)
  • Direct Media
This option tells the Asterisk server to never issue a reinvite to the client, if it is set to No. Select Yes if you want Asterisk to send reinvite to the client.
(Option button)
  • Qualify:
Timing interval in milliseconds at which a 'ping' is sent to a UAD/Phone or trunk in order to find out its status (online/offline).
In PBXware 5.1 'Qualify' is set to 8000 by default.
(ex. Set this option to '2500' to send a ping signal every 2.5 seconds to UAD/Phone or trunk. Navigate to 'Monitor: Extensions' or 'Monitor: Trunks' and check the 'Status' field)
([0-9])

Call Properties

Call Properties
  • Ringtime:
UAD/Phone ring time
(ex. Time in seconds that the UAD/Phone will ring before the call is considered unanswered)
([0-9])
  • Incoming Dial Options:
Advanced dial options for all incoming calls
(ex. Please see below for a detailed list of all available dial options( default: tr ))
([a-z])
  • Outgoing Dial Options:
Advanced dial options for all outgoing calls
(ex. Please see below for a detailed list of all available dial options( default: empty ))
([a-z])

Channels

Channels
  • Incoming Limit:
Maximum number of incoming calls
(ex. 2)
([0-9])
  • Outgoing Limit:
Maxmimum number of outgoing calls
(ex. 2)
([0-9])
  • Notransfer:
Disable native IAX transfer
(Option buttons)
  • Send ANI:
Should ANI ("super" Caller ID) be sent over this Provider
(ex. Set 'Yes' to enable)
(Option buttons)
  • Trunk:
Use IAX2 trunking with this host
(ex. Set 'Yes' to enable)
(Option buttons)

Authentication

Authentication
  • Auth Method:
Authentication method required by the provider
none
plaintext
md5
rc4
rsa
(ex. md5)
([a-z][0-9])

Codecs

Codecs
  • Disallow:
Set the codecs extension to 'not allowed'
(ex. This field is very unique. In order to work properly, this setting is automatically set to 'Disallow All' and it cannot be modified)
(Read only)
  • Allow:
Set the codecs extension is allowed to use
(ex. Only the codecs set under 'Settings: Server' will be available to choose from)
(Check box)

Auto Provisioning

Auto Provisioning
  • Auto Provisioning:
Enable auto provisioning service for this particular model.
(ex. Connect the UAD/Phone to PBXware without any hassle by providing the UAD/Phone MAC address (and optionally adding the Static UAD/Phone IP address and network details))
(Option buttons)
  • DHCP (Dynamic Hosts Configuration Protocol):
Set whether the UAD/Phone is on a DHCP or a Static IP address
(ex. Set DHCP = Yes if UAD/Phone is on dynamic or DHCP = No if UAD/Phone is on a static IP address. If on a static IP, you will have to provide more network details in the fields bellow).
(Option buttons)

Presence

Presence
  • Presence
Presence enables UAD to subscribe to hints which are used for BLF.
(ex. Yes, No, N/A)
  • User Agent Auto Provisioning Template:
This option is used for providing additional config parameters for SIP, IAX, and MGCP configuration files. Values provided here will be written into these configuration files.
([a-z][0-9])

DAHDI

General
  • Device Name:
Unique device name
(ex. AASTRA 480i)
([a-z][0-9])
  • Channels
Which card channels are used
(ex. 1,4/1-4)
  • Language:
Default language
([0-9][,-])
  • Status:
Extension status/presence on the network
(ex. If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select box by default when adding new extensions)
(Select box)
  • Signalling:
Signalling method
Example:
  • default
  • FXS Loopstart
  • FXS Groundstart
  • FXS Kewlstart
  • FXO Loopstart
  • FXO Groundstart
  • FXO Kewlstart
  • PRI CPE side
  • PRI Network side
(Select box) ,
  • Test Number:
This number is used for trunk monitoring in Monitor->Monitor Settings->Monitor Trunks.
  • Music On Hold:
Select which class of music to use for music on hold. If not specified the 'default' will be used
(ex. default)
(Select box)
  • Mailbox:
Define a voicemail context
(ex. 1234, 1234@context)
([a-z][0-9])
  • Group Method:
It specifies which group to dial, and how to choose a channel to use in the specified group. The four possible options are:
  • g: select the lowest-numbered non-busy DAHDI channel (aka. ascending sequential hunt group).
  • G: select the highest-numbered non-busy DAHDI channel (aka. descending sequential hunt group).
  • r: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group).
  • R: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group).

PRI

PRI
  • Switchtype:
Set switch type
Example:
  • National ISDN 2
  • Nortel DMS100
  • AT&T 4ESS
  • Lucent 5ESS
  • EuroISDN
  • Old National ISDN 1
(Select box)
  • PRI Dial Plan:
Set dial plan used by some switches
Example:
  • Unknown
  • Private ISDN
  • Local ISDN
  • National ISDN
  • International ISDN
(Select box)
  • PRI Local Dial Plan:
Set numbering dial plan for destinations called locally
  • Unknown
  • Private ISDN
  • Local ISDN
  • National ISDN
  • International ISDN
(Select box)
  • PRI Trust CID:
Trust provided caller id information
(ex. Yes, No, N/A)
(Option buttons)
  • PRI Indication:
How to report 'busy' and 'congestion' on a PRI
  • outofband - Signal Busy/Congestion out of band with RELEASE/DISCONNECT
  • inband - Signal Busy/Congestion using in-band tones
(Select box)
  • Network Specific Facility:
If required by switch, select network specific facility
  • none
  • sdn
  • megacom
  • accunet
(Select box)

Caller ID

Caller ID
  • Outbound Caller ID:
Caller ID set for all outbound calls where Caller ID is not set or supported by a device
(ex. john@domain.com)
([0-9])
  • Allow ES Caller ID:
CallerID can be set to 'asreceived' or a specific number if you want to override it.
NOTE: Caller ID can only be transmitted to the public phone network with supported hardware, such as a PRI. It is not possible to set external caller ID on analog lines
(ex. 'asreceived', 555648788)
([a-z][0-9])
  • Use Caller ID:
Whether or not to use caller ID
(ex. Yes, No, N/A)
(Option buttons)
  • Hide Caller ID:
Whether or not to hide outgoing caller ID
(ex. Yes, No, N/A)
(Option buttons)
  • Restrict CID
Whether or not to use the caller ID presentation for the outgoing call that the calling switch is sending
(ex. Yes, No, N/A)
(Option buttons)
  • Use CallerID Presentation:
Whether or not to use the caller ID presentation from the Asterisk channel for outgoing calls.
  • CID Signalling:
Set the type of caller ID signalling
Example:
  • bell - US
  • v23 - UK
  • dtmf - Denmark, Sweden and Netherlands
(Select box)
  • CID Start:
What signals the start of caller ID
Example:
  • ring = a ring signals the start
  • polarity = polarity reversal signals the start
(Select box)
  • Call Waiting CID:
Whether or not to enable call waiting on FXO lines
(ex. Yes, No, N/A)
(Option buttons)
  • Send CallerID After:
Some countries, like UK, have different ring tones (ring-ring), which means the caller id needs to be set later on, and not just after the first ring, as per the default.
(ex. Yes)
(Select box)

Echo Canceler

Echo Canceler
  • Echo Cancel:
Enable echo cancellation
(Yes, No, N/A)
(Option buttons)
  • Echo Training:
Mute the channel briefly, for 400ms, at the beginning of conversation, cancelling the echo. (Use this only if 'Echo Cancel' doesn't work as expected)
(ex. Yes, No, N/A)
(Option buttons)
  • Echo Cancel When Bridged:
Enable echo cancellation when bridged. Generally not necessary, and in fact undesirable, to echo cancel when the circuit path is entirely TDM
(ex. Yes, No, N/A)
(Option buttons)
  • Echo Canceller:
Types of echo cancellers:
  • mg2
  • oslec
  • kb1

Call Features

Call Features
  • Call Waiting:
Whether or not to enable call waiting on FXO lines
(ex. Yes, No, N/A)
(Option buttons)
  • Three Way Calling:
Support three-way calling. If enabled, call can be put on hold and you are able to make another call
(ex. Yes, No, N/A)
(Option buttons)
  • Transfer:
Support call transfer and also enable call parking (overrides the 'canpark' parameter). Requires 'Three Way Calling' = 'Yes'.
(ex. Yes, No, N/A)
(Option buttons)
  • Can Call Forward:
Support call forwarding
(ex. Yes, No, N/A)
(Option buttons)
  • Call Return:
Whether or not to support Call Return '*69'. Dials last caller extension number
(ex. Yes, No, N/A)
(Option buttons)
  • Overlap Dial:
Enable overlap dialing mode (sends overlap digits)
(ex. Yes, No, N/A)
(Option buttons)
  • Pulse Dial:
Use pulse dial instead of DTMF. Used by FXO (FXS signalling) devices
(ex. Yes, No, N/A)
(Option buttons)

Call Indications

Call Indications
  • Distinctive Ring Detection:
Whether or not to do distinctive ring detection on FXO lines
(ex. Yes, No, N/A)
(Option buttons)
  • Busy Detect:
Enable listening for the beep-beep busy pattern
(ex. Yes, No, N/A)
(Option buttons)
  • Busy Count:
How many busy tones to wait before hanging up. Bigger settings lower the probability of random hangups. 'Busy Detect' has to be enabled
Example:
  • 4
  • 6
  • 8
(Select box)
  • Call Progress:
Easily detect false hangups
(ex. Yes, No, N/A)
(Option buttons)
  • Immediate:
Should channel be answered immediately or the simple switch should provide dialtone, read digits, etc
(ex. Yes, No, N/A)
(Option buttons)

Call Groups

Call Groups
  • Call Group:
Set the Call Group to which the extension belongs.
(ex. Similar to 'Context' grouping, only this option sets to which call group extension belongs(Allowed range 0-63))
([0-9][,-])
  • Pickup Group:
Set groups extension as allowed to pickup.
(ex. Similar to 'Context' grouping, only this option sets the Call Groups extension as allowed to pickup by dialing '*8'.)
([0-9][,-])

RX/TX

RX/TX
  • RX Wink:
Set timing parameters
Example:
  • Pre-wink (50ms)
  • Pre-flash (50ms)
  • Wink (150ms)
  • Receiver flashtime (250ms)
  • Receiver wink (300ms)
  • Debounce timing (600ms)
(Select box)
  • RX Gain:
Receive signal decibel
(ex. 2)
([0-9])
  • TX Gain:
Transmit signal decibel
(ex. 2)
([0-9])

Other DAHDI Options

Other Zapata Options
  • ADSI (Analog Display Services Interface):
Enable remote control of screen phone with softkeys. (Only if you have ADSI compatible CPE equipment)
(ex. Yes, No, N/A)
(Option buttons)
  • Jitter Buffers:
Configure jitter buffers. Each one is 20ms long
(ex. 4)
([0-9])
  • Relax DTMF:
If you are having trouble with DTMF detection, you can relax the DTMF detection parameters
(ex. Yes, No, N/A)
(Option buttons)
  • Fax Detect:
Enable fax detection
Example:
  • both
  • incoming
  • outgoing
  • no
(Select box)

Span

Span
  • Span number:
Number of the span
(ex. 1)
([0-9])
  • Span timing:
How to synchronize the timing devices
Example:
  • 0 - do not use this span as a sync source
  • 1 - use as the primary sync source
  • 2 - set as the secondary, and so forth
([a-z])
  • Line build out:
Length of the last leg of the connection and is set to zero if the length is less than 133 feet
Example:
  • 0 db (CSU) / 0-133 feet (DSX-1)
  • 133-266 feet (DSX-1)
  • 266-399 feet (DSX-1)
  • 399-533 feet (DSX-1)
  • 533-655 feet (DSX-1)
  • -7.5db (CSU)
  • -15db (CSU)
  • -22.5db (CSU)
(Select box)
  • Framing:
How to communicate with the hardware at the other end of the line
Example:
  • For T1: Framing is one of d4 or esf.
  • For E1: Framing in one of cas or ccs.
(Select box)
  • Coding:
How to encode the communication with the other end of line hardware.
Example:
  • For T1: coding is one of ami or b8zs
  • For E1: coding is one of ami or hdb3 (E1 may also need crc)
(Select box)
  • Yellow:
Whether the yellow alarm is transmitted when no channels are open.
(ex. Yes, No, N/A)
(Option buttons)

Dynamic Span

Dynamic Span
  • Dynamic span driver:
The name of the driver (e.g. eth)
([0-9][a-z])
  • Dynamic span address:
Driver specific address (like a MAC for eth).
([0-9][a-z])
  • Dynamic span channels:
Number of channels.
(6)
  • Dynamic span timing:
Sets timing priority, like for a normal span. Use "0" in order not to use this as a timing source, or prioritize them as primary, secondary, etc.
(0)

FXO Channels

FXO Channels
  • FXO Loopstart:
Channel(s) are signalled using FXO Loopstart protocol
([0-9])
  • FXO Groundstart:
Channel(s) are signalled using FXO Groundstart protocol
([0-9])
  • FXO Kewlstart:
Channel(s) are signalled using FXO Kewlstart protocol
([0-9])

Values for above fields are set as follows:

  • 1 - for one card
  • 1-2 - for two cards
  • 1-3 - for three cards etc
  • 2-3 (If your card has modules in this order FXS, FXO, FXO, FXS)

FXS Channels

FXS Channels
  • FXS Loopstart:
Channel(s) are signalled using FXS Loopstart protocol
([0-9])
  • FXS Groundstart:
Channel(s) are signalled using FXS Groundstart protocol
([0-9])
  • FXS Kewlstart:
Channel(s) are signalled using FXS Kewlstart protocol
([0-9])

Values for the above fields are set as follows:

  • 1 - for one card
  • 1-2 - for two cards
  • 1-3 - for three cards etc
  • 2-3 (If your card has modules in this order FXS, FXO, FXO, FXS)

Jitter Buffer

Jitter Buffer

A jitter buffer temporarily stores arriving packets in order to minimize delay variations. If packets arrive too late then they are discarded. A jitter buffer may be mis-configured and be either too large or too small.

  • Enable Jitterbuffer:
Enables the use of a jitterbuffer on the receiving side of a SIP channel. An enabled jitterbuffer will be used only if the sending side can create and the receiving side cannot accept jitter. The SIP channel can accept jitter, thus a jitterbuffer on the receive SIP side will be used only if it is forced and enabled.
(ex. Yes, No, N/A)
(Option buttons)
  • Force Jitterbuffer:
Should we force jitter buffer (default value 10)
(ex. Jitter buffer is usually handled by the UADs/Phones. But in case these do this poorly , the jitter buffer can be enforced on PBXware side)
([0-9])
  • Max length (ms):
Max length of the jitterbuffer in milliseconds.
(ex. 300)
([0-9])
  • Re-sync threshold:
Resync the threshold for noticing a change in delay measured
(ex. 1000)
([0-9])
  • Implementation:
Jitter buffer implementation used on a receiving side of a SIP channel. Defaults to "fixed".
(ex. adaptive)
(Select box)
  • Target extra
  • Logging:
Enables jitterbuffer frame logging. Defaults to "no".
(ex. Yes, No, N/A)
(Option buttons)

Other DAHDI Channels

Other DAHDI Channels
  • Unused:
([0-9])
  • Clear:
([0-9])

Access Codes

Access codes provide tenant user with access to essential system or enhanced services

Voicemail

Access Codes
  • Voicemail:
Voice inbox access code. This number is dialed to access the extension voice inbox (extension PIN required)
(ex. From extension 1000, dial '*123' to access the extension 1000 voice inbox. When asked for a PIN, provide the PIN set for this extension)
([0-9])
  • General Voicemail:
Access code for general voice mailbox. This number is used for checking your voice inbox from any system Extension
(ex. Dial '*124'. Enter your Extension number and PIN when asked for it)
([0-9])
  • Voicemail Transfer:
Access code for transferring active calls to any system voice box
(ex. During active conversation dial '*125 + $EXTENSION' to transfer calling party to system $EXTENSION number voice box)
([0-9])

Enhanced Services

  • Last Caller:
Access code for dialing the last Extension that called you
(ex. If Extension '1000' was the last extension that called you, dial '*149' from your Extension. Message 'The number to call your line was $EXTENSION. To call this number press 1' will be played. Press '1' dial this destination).
([0-9])
  • Monitoring:
Access code for monitoring active calls
(ex. If Extensions '1000' and '1001' are in conversation, dial '*199 + 1000' to listen the ongoing conversation)
([0-9])

Speakerphone Paging

  • Speakerphone Page:
Access code for transmitting a message to multiple phones via their loudspeakers
(ex. Dial '*399' to speakerphone all Extensions defined under Enhanced Services)
([0-9])
  • Single Speakerphone Page:
Access code for transmitting a message to phone loudspeaker
(ex. Dial '*400 + $EXTENSION' to transmit a message to provided extension (phone) loudspeaker)
([0-9])
  • Groups Speakerphone Page:
Access code for transmitting a message to paging groups via their loudspeakers
(ex. Dial '*600 + Paging Group' to transmit a message over intercom to all the extensions added to Paging group)
([0-9])

Features

  • Speed Dial:
Speed Dial enables faster dialing by entering access code and dual number speed dialing code. Speed Dialing Codes can be entered by clicking Settings: Servers and then clicking on Speed Dial button
(ex. Dial '*130 + XX' where XX represents speed dialing code associated to some extension)
([0-9])
  • Other Networks:
Access code for accessing other PBXware MT networks
Example:
Dial '*188 + $NETWORK + $EXTENSION' to dial number $EXTENSION or $NETWORK (e.g. '*188 8 1000').
NOTE: 'Trunk' and 'Other Network' must already be set
([0-9])
  • Listen to CDR recordings:
Access code used when you want to listen last 9 Call Recordings from CDRs
(ex. Dial '*170' access code followed by a number between 1 and 9, where 1 is most recent recorded conversation (excluding access code calls)).
([0-9])

Call Forwarding

  • Enable Call Forwarding:
Access code for enabling Call Forwarding Enhanced Service
(ex. Dial '*71 + $EXTENSION' to forward all calls to $EXTENSION number. This number can be local Extension or Proper/Mobile number (e.g. '*71 1001' or '*71 55510205'))
([0-9])
  • Disable Call Forwarding:
Access code for disabling Call Forwarding Enhanced Service
(ex. Dial '*72' to disable this service (Extension number not required))
([0-9])

Group Hunt

  • Enable Do Not Disturb:
Access code for enabling Do Not Disturb option
(ex. Dial *71 to enable this service (Extension number not required)
([0-9])
  • Disable Do Not Disturb:
Access code for disabling Do Not Disturb (DND) option
(ex. Dial *72 to disable this service, extension number is not required.
([0-9])
  • Enable Group Hunt:
Access code for enabling Group Hunt option
(ex. Dial *510 to enable this option, extension number not required
([0-9])
  • Disable Group Hunt:
Access code for disabling Group Hunt option
(ex. Dial *511 to disable this option, extension number is not required)
([0-9])

Caller ID

  • Block CallerID:
Access code to block other users from seeing your CallerID
(ex. Dial '*67' to block displaying your CallerID)
([0-9])
  • Block CallerID once:
Access code for blocking only first next call from displaying CallerID
(ex. Dial '*81' to block only next call from seeing your CallerID)
([0-9])
  • Unblock CallerID:
Access code to unblock your CallerID after it has been blocked
(ex. Dial '*68' to unblock your CallerID)
([0-9])
  • Call with CallerID list number:
Access code for changing caller ID number
(ex. Dial *65*[Caller ID number as entered in the list]*[Number you wish to dial])

Call Parking

  • Call Park:
Access code for parking active calls
(ex. During active conversation dial '#700'. The call will be parked on first available Call Park Extension (e.g. 701). A parked call can be picked up from any system Extension by dialing parked Extension ('701')
NOTE: Extensions must have call pickup enabled in Enhanced Services in order to be able to pick up parked calls.
([0-9])
  • Call Park Start:
Start Extension for call parking service
(ex. If set to '701' all calls will be parked on Extensions '701' to 'Call Park End')
([0-9])
  • Call Park End:
End Extension for call parking service
(ex. If set to '720' all calls will be parked on Extensions 'Call Park Start' to '720')
([0-9])
  • Call Park Timeout (sec):
Default timeout for Call Park is 45 seconds which means that once the call is parked it will stay at the assigned parking lot for 45 seconds after which call will return to an extension which it was parked from.
  • Enhanced Call Park:
Default Access code for Enhanced Call park is *755. Once used on live call this feature will transfer call to Announce Extension. Timeout, Announce Extension and Timeout extension can be set under Settings -> Tenants -> TenantName -> Enhanced Call Parking (section).
For example, while on active call you can dial '#800'. The call will be parked and Announce Extension will ring for set number of seconds. After that period call will be directed to Timeout Extension.
([0-9])
  • Call Pickup:
Default Access code for Call pickup is *88+EXT_NUM.
Call Pickup feature enables users to pick up calls that are ringing other extensions in associated call groups. Along with call pickup enabling you to pick up calls that are ringing any extension in your call group by simply dialing *8, it is possible to use directed pickup and pick up calls from a specific extension by dialing *88 + EXTENSION NUMBER.
([0-9])
NOTE: The Call Pickup access code will be applied to phones if auto provisioning is used.
  • Call Pickup Asterisk:
Default Access code for Call pickup is *8 and
  • Parked Calls Transfer:
Set this feature as Disable (default setting) or Enable it for both sides or for the pickup side only.

System Tests

  • Music On Hold:
Access code for playing Music On Hold sound files
(ex. Dial '*388' to play 'default' Music On Hold class sound files)
([0-9])
  • Echo Audio Read:
Access code for echo audio test
(ex. Dial '*398' and talk. Everything you say is returned back so the server response time can be checked)
([0-9])
NOTE: The user will be able to skip the audio prompt when doing Echo Test by pressing any DTMF, including "#".

Greetings

  • Record Greeting:
Record greeting or any other sound file that can be used in PBXware
(ex. Dial '*301' and after the beep say the message that you want to record)
([0-9])
  • Agent Greeting:
Record the agent greeting which the caller will hear before being connected to an agent
(ex. Dial *302 and then enter the agent number to record the agent greeting which the caller will hear before being connected to an agent)
([0-9])
  • Change Greeting (greeting access code for specific destination types):
(e.g., Dial *303xxx where xxx is an IVR, Ring Group, Queue, etc. When dialed the user is prompted to record a new greeting. Once accepted, the greeting is set as the destination's new greeting.)
  • Overwrite Greeting (greeting access code for specific destination types):
(e.g., Dial *304xxx where xxx is an IVR, Ring Group, Queue, etc. When dialed the user is prompted to record a new greeting. Once accepted, the greeting is set as the destination's new greeting and the old greeting, if it existed, will be deleted.)

Operation Times

  • Open Operation Times:
User will dial '*401' to open systems operation times
(ex. If operation times is open, but not explicitly closed, it is automatically closed at 12 AM)
([0-9])
  • Close Operation Times:
User will dial '*402' to close systems operation times
(ex. If user doesn't close operation times, it will be closed automatically at 12 AM)
([0-9])
  • Reset Operation Times:
User will dial '*403' to reset systems operation times and restore rules entered in settings.
  • Midnight Reset (5.3 update):
Users have the ability to set the operation times midnight reset option to Yes/No/Not Set.

By default this option is set to "Yes", meaning that the access codes will be reset at midnight. If set to "No", midnight reset will be skipped.

Midnight Reset




Operation Times BLFs

  • Open/Close: *404
  • Close/reset: *405
Users can program a BLF key on their phone to *404 in order to monitor Operation Times status and to use it to enable or disable Operation Times rules.
For example, if a user stays in the office after the Operation Times rules activate for off hours, the BLF will notify the user of the changed state by turning on the BLF light or changing its color (behavior will depend on the device used). Once *404 is provisioned as BLF pressing the button will toggle Operation times ON or OFF, the same as if user dialed *401/*402. To reset Operation Times rules, user will have to dial *405 manually.

NOTE: The System operation times ES has to be enabled for this to work.

Follow Me

  • Enable Follow Me:
Access Code for enabling "Follow Me" Enhanced Service.
Dial *520 to enable "Follow Me" and use set of rules defined for this service in extensions Enhanced Services.
(*[0-9])
  • Disable Follow Me:
Access Code for disabling "Follow Me" Enhanced Service.
Dial *521 to disable "Follow Me" and use set of rules defined for this service in extensions Enhanced Services.
(*[0-9])

Hot Desking

  • Hot Desking:
Access code for enabling/disabling Hot Desking.
(ex. Dial *555 to enable/disable Hotdesking)
([0-9])

Parking Lots

Parking Lots is new feature introduced in Multi Tenant version 4.1. It allows users to park a call to specific parking lot.

Parking Lots
  • Parking Lot:
Name of the parking lot.
  • Parking Extension:
Number of parking extension.
  • Parking Positions:
Number of parking position, or number range in form of XXX-XXX
  • Last Destination:
Number of extension which represents Last Destination. After call is parked, if no one pick up the call, call will go to last destination extension, after time specified in Timeout field exceeds.

Add Parking Lot

Add Parking Lot
  • Name:
Enter the name of Parking Lot.
  • Parking Extension:
Enter the number of parking extension. If user wants to park the call, user will transfer the call to extension entered in this field.
  • Parking Positions:
Enter the number range in form of XXX-XXX. After the call is parked, to pick up the call, user needs to dial one of the number listed here.
  • Timeout [s]:
Timeout is time in second. During this time period, call can be picked. After timeout exceeds, the call will be redirected to the Timeout Destination.
  • Timeout Destination:
Enter the Timeout Destination. This extension will be dialed if no one picked the parked call during the time specified in Timeout field.
  • Custom Ring Tone:
If you are directing calls to an extension on which a supported UAD is registered, you could set a Custom Ringtone with which the phone will ring. Read more on this in the Custom Ring Tones - HOWTO
(ex. <Simple-3>)
([a-z][0-9])
  • Caller ID Prefix:
Set the Caller ID Prefix. This caller ID prefix will be set when call is redirected to Timeout Destination.

Numbering Defaults

Numbering defaults sets the way PBXware will assign network numbers to Extensions, Conferences, etc.

Numbering Defaults

Fetch least unallocated number (default):

This option takes the least number on the system that is not used by the system and assigns it to a new Extension you're trying to create, for example.

Fetch next unallocated number:

If the last allocated number was assigned to extension 2010, the next IVR that you're trying to add, for example, will be given number 2011. The system will not try to give you an unallocated number 1022, for example.

Fetch random:

Use this option in case you want to assign numbers in a non-sequential order. If you create a new Queue, for example, PBXware will assign it a number 2350, for example. And if you try to create a new Extension right after that, PBXware might assign it a number 9838, for example.


E-mail templates

E-mail templates
Voicemail E-mail Template


Set the e-mail template used for sending emails when new Voicemail is received.


Speed Dial

Speed Dial is used with the *130 Access Code. When you dial *130XX, where XX is a two digit Speed Dial Code, you will dial the extension associated with that code.

The difference here is that you set Speed Dial Codes for all extensions on the system, not just the ones which have this turned on in ES

TIP

This is useful only if you have more than 6 digits in your extensions.

Speed Dial
  • Code (XXX)
Three digit code which is entered after the Speed Dial Access Code, *130 as default
(ex. 22)
([0-9])
  • Speed Dial Name
Short description of the Destination to which this Code points.
(ex. Sales-John)
([a-z][0-9])
  • Destination
Destination to which this Code is pointing.
(ex. 1005)
([0-9])

CSV Upload is used when you have all the codes written in a simple CSV file in form:

  • Code,Name,Destination

CSV Download is used when you want to download already set Dial Codes in CSV file

CLI Routing

  • CLI Routing and validation per tenant (tenant administrator can manage validation and routing for tenant).

This option is used to fine-tune the functionality by adding rules which send the calls to different destinations based on Caller IDs.

Example:

If the caller's ID begins with 203, you can send all those calls to the IVR, if the CallerID is a specific number such as 063456789, then send that call to extension 4444.
CLI Routing
  • CSV upload (CSV upload and CSV template added)
  • Search (implemented Search for available routes)
CLI Routing Search

About

About section displays the system's edition, version, release date and licensing information.

About
  • PBXware MT:
This line identifies the PBXware MT version, release date (Revision) and Asterisk running
(ex. Edition: Multi-Tenant Release: 5.1 (ba2ce051) Running: 13.16.0-dev-gc-9f9ba607)
(Display)
  • Package:
Package information displays a number of system Extensions, Trunks, Conferences, etc.
(ex. Extensions 768, Voicemails 100...)
(Display)


Next -> 21.Admin Settings