MT 3.8.5 Settings

From Bicom Systems Wiki

Settings

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'


Settings: On master tenant






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)
  • Package:
Name of the package used by a slave tenant.
(ex. Full package)
(Display)
  • edit.png Edit
Edits the tenant configuration
(ex. Click to edit tenant configuration)
(Button)
  • delete.png Delete
Deletes a tenant from the system
(ex. Click to delete a tenant from the system)
(Button)


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])
  • snap0094.gif Network details configuration window
(ex. Click to edit network configuration)
(Button)

Server Details

Server Details
  • Root Password:
Set the root password
(ex. PBXware MT prompts for this password during the system/ssh login and when accessing system services through the interface)
([a-z][0-9])
  • Confirm Password:
Confirm the root password
(ex. Re-type the Root Password entered in the field above)
([a-z][0-9])
  • Time zone:
Time zone in which PBXware MT is located
(ex. Select the appropriate time zone, for example 'USA/East-coast)
(Select box)
  • Hostname:
The name given to the machine which will identify the system on the network
(ex. hostname)
([a-z][0-9])

General Settings

General 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)
  • 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)
  • 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)
  • Default domain for OSC login:
This option allows you to enter default e-mail domain that will be used in case you would like to login to OSC using username only.
For example, if you add bicomsystems.com in Default domain for OS login, this would allow you to enter support as a username, instead support@bicomsystems.com when logging in On-line Self Care. Users that have their accounts associated with e-mail address not on bicomsystems.com would have to enter full e-mail address.
([a-z][0-9])
  • Disable Tenant to Tenant calls:
When this option is enabled, extensions from one tenant will not 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)

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.

IVR 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 when all calls are redirected to 'Default Destination'. If 'Destination' field in the Closed dates is set, call will not go to '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.

Call Recordings

Call Recordings
  • Record calls by default:
When creating an extension, Yes, No or N/A will be selected by default, depending on the selection that you make here
(ex. Yes, No, N/A)
(Option buttons)
  • Use MixMonitor:
MixMonitor allows you to record conversations with the possibility to adjust the heard and spoken volume and to append the next conversation in the same file. So, at the end of the day, you could have all the conversations on one channel in one file. They will be stored in the same sequence, as they are made.
(ex. Yes, No, N/A)
(Option buttons)
  • Silent recording by default:
When creating an extension, Yes, No, or N/A will be selected for Silent recording, depending on the selection you make here
(ex. Yes, No, N/A)
(Option buttons)
  • Email instant recording interval (min):
How often to send an email of instant recording in minutes. (Only on master tenant)
(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

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
(ex. Yes, No, N/A)
(Option buttons)

Other Networks

The system can be part of the 'default' PBXware MT network where all extensions share the same unified dial plan. This is achieved by selecting from the select box to which the PBXware MT network system belongs. Clicking on 'Other network' will open the following options:

Other Networks
  • Mode:
Sets the way other PBXware MT 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
(ex. Network London)
(Display)
  • Number:
Other network access number
(ex. 8)
(Display)
  • Trunk:
Trunk used once the other network number is dialed
(ex. 2554433)
(Display)
  • edit.png Edit
Edits an other network configuration
(ex. Click to edit an other network configuration)
(Button)
  • delete.png Delete
Deletes an other network connection
(ex. Click to delete an other network connection from the system)
(Button)
Add/Edit Network
Add/Edit Network
  • Name:
Custom Other Network name/identifier
(ex. London FO/7)
([a-z][0-9])
  • Prefix:
Number used to access the Other Network (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 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 Other Network Access Code
(ex. If this field is set to '2', only extensions on Other Network starting with number 2 will be allowed to dial. If you dial *188 7 1002, our call will fail. But, if we dial *188 7 2000, our 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 provider requires a certain code before dialing the destination, area code inserted automatically etc...)
([a-z][0-9])
  • Trunk:
Select trunk that will be used once 'Number' is dialed
(ex. London FO Trunk)
(Select box)

Speed Dial

Speed Dial is used with *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 the 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 then 6 digits in your extensions.

Speed Dial
  • Code (XX)
Two 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 simple CSV file in the following form:

  • Code,Name,Destination

CSV Download is used when you want to download already set Dial Codes in a 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. (Only on master tenant)

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

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.

Email
  • 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)
  • Zaptel Zone:
Overrides Automatic Country Detection and this is set on the master tenant where slaves are using this value.
(ex. It is recommended to keep this setting always set to 'Automatic')
(Select box)
  • Indications:
Which indications (Ringing, Busy, etc. sounds) are to be used by the system, also defined on master tenant.
(ex. If the system is located in USA, set 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 striped 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

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
  • 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.

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])
  • Zaptel:
Total number of all system trunks using ZAPTEL 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.

Numbering Defaults

Numbering default is set during the initial system set up in order to set how many digits the system will use as default.

Numbering Defaults
  • Number of Digits:
Number of digits used by the system to create local extensions, IVRs, Queues, Voicemail boxes, Conferences etc.
(ex. This option is available for settings only during the setup wizard install process. In order to change the number of digits after the setup wizard, please remove all Extensions, DIDs, Conferences (all apps with network number). The recommended value for this field is 4).
(Select box)
  • PSTN numbering mode:
If this option is turned on, when a number is dialed there is no need for DID, the system calls that extension automatically.
(ex. Yes, No, N/A)
(Option buttons)
NOTE: This is only applicable for 10-digit numbering systems where all extensions start with 1.
  • Call groups/Pickup groups:
Select which Call group/Pickup group tenant could use.
(ex. Ctrl+Left Mouse Click to select multiple groups).
(Select box)

Enhanced call parking

Configure options which affect the Enhanced Call Parking feature of PBXware.

Enhanced Call Parking
  • Timeout (sec):
Define how long the call will stay parked before it is being returned to the Timeout Extension
(ex. 5)
([0-9])
  • Announce Extension:
Which extension is going to ring when the call is being parked using Enhanced Call Parking
(ex. 1001)
([0-9])
  • Timeout Extension:
If the Timeout expires and Announce Extension doesn't answer the call which was parked, that call will go to this extension
(ex. 1002)
([0-9])

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)


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])

DynDNS Update Service

DynDNS Update Service
  • Run every:
Select the time interval at which the DynDNS update request will be sent
(ex. Select 'minute' to send request to the DynDNS server every minute)
(Select box)
  • Mode:
Set the way the DynDNS service will work on the system
(ex. 'Update & Monitor' will monitor and update/reload the IP address(es) of monitored servers once they change. 'Monitor only' will monitor for IP address change but will not update/reload)
(Select box)
  • Dynamic Hostname:
Hostname to which requests are periodically sent
(ex. domain.dyndns.org)
([a-z][0-9])
  • Username:
Username for DynDNS authentication
(ex. Type the username used for DynDNS authentication here)
([a-z][0-9])
  • Password:
Password for DynDNS authentication
(ex. Type the password used for DynDNS authentication here)
([a-z][0-9])
  • DynDNS Server:
DynDNS server name
(ex. members.dyndns.org, (default value))
([a-z][0-9])
  • CheckIP Server:
DynDNS server for checking the IP addresses
(ex. checkip.dyndns.org, (default value))
([a-z][0-9])
  • Monitor Servers:
Dyndns domains of monitored servers separated by a blank space
(ex. serverone.dyndnd.org)
([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])


Protocols

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

Supported protocols are:

  • SIP
  • IAX
  • WOOMERA

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])
  • 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])
  • SRV lookup:
Enable DNS SRV lookups on outbound calls
(ex. Disabling this option will disable SIP calls based on domain names between SIP users on the Internet)
(Option buttons)
  • 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 Overlap Dialing:
With overlap dial set to no, the gateway expects to receive the digits one right after the other coming in to this line with very little delay between digits. With overlap dial set to yes, then the device waits up to about 2 seconds between digits.
(ex. Yes, No, N/A)
(Option buttons)
  • Allow SIP Transfers:
Disable all transfers (unless enabled in peers or users). Default is enabled.
(ex. Yes, No, N/A)
(Option buttons)
  • 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 and TLS Settings
  • Enable TCP:
Enable server for incoming TCP connections. (Default value: No)
ex. Yes, No, N/A
  • Authentication Timeout:
Authentication timeout specifies the maximum number of seconds a client has to authenticate. If the client does not authenticate before this timeout expires, the client will be disconnected. (default: 30 seconds)
  • Authentication limit:
Authentication timeout specifies the maximum number of unauthenticated sessions that will be allowed to connect at any given time. (default: 100)

TLS Settings

  • Enable TLS:
Enable server for incoming TLS (secure) connections.
Options: Yes, No, N/A (default is no).

NAT

NAT
  • External IP:
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 TCP port:
The externally mapped TCP port when Asterisk is behind a static NAT or PAT.
External TCP Port will default to the External IP or External host port if either one is set.
  • External TLS port:
The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
External TLS port will default to the RFC designated port of 5061.
  • External Host (DynDNS):
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])
  • External Host Refresh:
How often to refresh the External DynDNS host (if used)
(ex. Time in seconds (e.g. 10))
([0-9])
  • Match External IP Locally:
Only substitute the external IP or external host setting if it matches your local network setting.
(ex. Yes, No, N/A)
(Option buttons)
  • Local network:
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
(ex. Set this option to 'Yes' if the system is behind NAT)
(Option buttons)

Security

Security
  • Always Reject with 401:
Select whether PBXware will reject the INVITE or REGISTER with 401 Unauthorized message without letting the caller know if there was a matching user or peer
(Option buttons)
  • Allow guest:
Define whether PBXware will accept guest connections without authorization if the end device doesn't support authorization
(Option buttons)
  • Allow External INVITEs:
If set to no, this option will disable INVITE and REFER messages to non-local domains
(Option buttons)
  • Allow REDIR: Allows support for 302 Redirect, and will redirect them all to the local extension returned in Contact rather than to that extension at the destination)
(Option buttons)
  • Authenticate OPTION 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

  • Minimum roundtrip time for monitored hosts:
Minimum roundtrip time for messages to monitored hosts. Defaults to 100 ms
  • Default T1 Timer:
Default T1 timer.
Defaults to 500 ms or the measured round-trip time to a peer (qualify=yes).
  • Call Setup Timer
SIP Session Timers
Call setup timer.
If a provisional response is not received in this amount of time, the call will autocongest. Defaults to 64*timert1.

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
  • Trust Remote-Party-ID:
Defines whether PBXware allow Remote-Party-ID header
(Option button)
  • Send Remote-Party-ID:
Should 'Remote-Party-ID' be added to uri.
Options:
  • Use Remote-Party-Id - Use the "Remote-Party-ID" header to send the identity of the remote party
  • Use P-Asserted-Identity - Use the "P-Asserted-Identity" header to send the identity of the remote party
(Select Box)
  • RPID with SIP UPDATE:
In certain cases, the only method by which a connected line change may be immediately transmitted is with a SIP UPDATE request. If communicating with another Asterisk server, and you wish to be able transmit such UPDATE messages to it, then you must enable this option. Otherwise, we will have to wait until we can send a reinvite to transmit the information.

Misc

Misc
  • Pedantic checking:
Enable slow, pedantic checking for Pingtel and multi-line formatted headers for strict SIP compatibility
(ex. It is recommended to set this field to 'No')
(Option buttons)
  • 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|)
  • Premature Media
This option enables or disables Premature Media.
Some ISDN links send empty media frames before the call is in ringing or progress state. The SIP channel will then send 183 indicating early media which will be empty - thus users get no ring signal.Setting this to "yes" will stop any media before we have call progress (meaning the SIP channel will not send 183 Session Progress for early media). Default is "yes". Also make sure that the SIP peer is configured with progressinband=never. In order for "noanswer" applications to work, you need to run the progress() application in the priority before the app.
Options: Yes, No, N/A
  • 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)
  • Add ;user=phone:
Should ';user=phone' be added ot uri
(ex. You're recommended to set this option to 'No' unless required otherwise)
(Option buttons)
  • 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)
  • Manager events on SIP events:
Should manager events be generated if SIP UAD/Phone performs some event (Hold for example)
(ex. You're recommended to set this option to 'No' unless required otherwise)
(Option buttons)
  • Shrink CallerID
This option enables or disables Shrink CallerID.
Options: Yes, No, N/A
NOTE: The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not in square brackets. For example, the caller id value 555.5555 becomes 5555555 when this option is enabled. Disabling this option results in no modification of the caller ID value, which is necessary when the caller id represents something that must be preserved. This option by default is turned on.
  • Set Reason header:
Set this option to yes to add Reason header and use Reason header if it is available.
Options: Yes, No, N/A

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])
  • Auth debugging:
Should authentication be debugged
(ex. Setting this option to 'Yes' will increase the amount of debugging traffic)
(Option buttons)
  • Match Authentication username
If available, match user entry using the 'username' field from the authentication line instead of the From: field.
  • Generate failure events in Manager
If this option is set to yes, asterisk will generate manager "peerstatus" events when peer can't authenticate with Asterisk. Peerstatus will be "rejected".
Default value: No
Options: Yes, No, N/A

Registration

Registration
  • Length of i/o reg:
Max duration (in seconds) of incoming registration we allow
(ex. 7200)
([0-9])
  • Def. Length of i/o reg:
Default duration (in seconds) of incoming/outgoing registration
(ex. 3600)
([0-9])
  • Registration context:
Should system dynamically create and destroy noop priority 1 extension for peer who (un)registers with us
(ex. sipregistrations)
([a-z][0-9])
  • Registration timeout:
Number of seconds after which registration times out
(ex. Default value 20)
([0-9])
  • Register attempts:
Number of registration attemps
(ex. One 'Register timeout' equals one 'Registration attempts'. Default value 10)
([0-9])

Qualification

Registration
  • Qualify:
Timing interval in milliseconds at which a 'ping' is sent to a host in order to find out its status
(ex. Set this field to 2000, for example. If more time than provided here is needed to reach the host, the host is considered offline)
([0-9])
  • Qualify Frequency
How often to check for the host to be up in seconds
Set to low value if you use low timeout for NAT of UDP sessions.
(|0-9|)
  • Qualify groups
Number of peers in a group to be qualified at the same time
Default: 1
(|0-9|)
  • Qualify groups gap
Number of milliseconds between each group of peers being qualified
Default: 100
(|0-9|)

MWI

MWI
  • MWI Mime-type:
Allow overriding of mime type
(ex. Default value 'text/plain')
([a-z])
  • Check MWI time:
Default time between mailbox checks for peers
(ex. Default value 10)
([0-9])
  • Voicemail extension:
Dial plan extension to reach the mailbox. This option sets the 'Message-Account' in the MWI notify message
(ex. Default value 'asterisk')
([a-z][0-9])
  • 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|)
  • Enable Buggy MWI
Cisco SIP firmware doesn't support the MWI RFC fully. Enable this option to not get error messages when sending MWI to phones with this bug.
Options: Yes, No, N/A
(Option buttons)

Subscriptions

Subscriptions
  • Allow Subscriptions:
Disable support for subscriptions. Default is Yes.
(ex. Yes, No, N/A)
(Option buttons)
  • Subscribe Context:
Set a specific context for SUBSCRIBE requests.
This option is useful to limit subscriptions to local extensions.
([a-z][0-9])
  • Notify on RINGING:
Notify subscriptions on RINGING state
(ex. Yes, No, N/A)
(Option buttons)
  • Notify on HOLD:
Disable support for subscriptions. Default is No.
(ex. Yes, No, N/A)
(Option buttons)

Domains

Domains
  • Domain:
Set default domain for this host
If configured, Asterisk will only allow INVITE and REFER to non-local domains. Use 'sip show domains' in asterisk CLI to list local domains.
([a-z][0-9])
  • Auto Domain:
Turn this on to have Asterisk add a local host name and a local IP to domain list.
If system host name is set to 'my_system', with this feature set to 'Yes', 'my_system' will be automatically added to domain list. Options: Yes, No, N/A
(Option buttons)
  • From Domain:
Change the 'From: ' headers
(ex. Keep this field empty unless requested otherwise)
([a-z][0-9])
  • Allow External Domains:
Should domains not serviced by this server be allowed.
Options: Yes, No, N/A
(Option buttons)

Codecs

Codecs
  • Disallow:
Set the codecs extension to not allowed to use
(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 to 'allowed to use'
(ex. Only the codecs set under 'Settings: Server' will be available to choose from)
(Check box)


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.


  • Preffered Codecs only
(ex. Yes, No, N/A)
  • Auto-Framing (RTP Packetization):
If auto framing is turned on, system will choose packetization level based on remote ends preferences.
(ex. Yes, No, N/A)
(Option buttons)
  • Non-standard G726:
If the peer negotiates G726-32 audio, use AAL2 packing order instead of RFC3551 packing order.
(ex. Yes, No, N/A)

Video support

Video Support section
  • Video support
This option enables or disables PBXware video support.
Options: Enable, Disable and Always.
  • Max Call Bitrate:
Maximum bitrate for video calls (default 384 kb/s)


NOTE:Video Support and Max Call Bitrate can be set for peers and users as well.

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.

Jitter Buffer
  • 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)
  • Logging:
Enables jitterbuffer frame logging. Defaults to "no".
(ex. Yes, No, N/A)
(Option buttons)

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.
  • Enabled (FEC, ECC) - Enables T.38 with FEC error correction.
  • Enabled(FEC,Maxdatagram=400) - Enables T.38 with FEC error correction and overrides the other endpoint's provided value to assume we can send 400 byte T.38 FAX packets to it.
  • Enabled (Redundancy ECC) - Enables T.38 with redundancy error correction.
  • Enabled (No ECC) - Enables T.38 with no error correction.

NOTE: In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that is based on an incorrect interpretation of the T.38 recommendation, and results in failures because Asterisk does not believe it can send T.38 packets of a reasonable size to that endpoint (Cisco media gateways are one example of this situation). In these cases, during a T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you can set an override (globally, or on a per-device basis) to make Asterisk ignore the T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead. This can be done by selecting Enabled (FEC,Maxdatagram=400) option.

  • Ignore version (OCS)
By default, Asterisk will honor the session version number in SDP packets and will only modify the SDP session if the version number changes. Setting this option to yes will force asterisk to ignore the SDP session version number and treat all SDP data as new data. This is required for devices that send us non standard SDP packets. By default this option is off.
(Yes, No, N/A)
  • 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])

SIP Debugging

Adjust options that affect SIP debugging on the system

SIP Debugging
  • SIP Debug:
Should SIP Debug be turned on all the time
(ex. It is recommended to set this option to 'No' unless required otherwise)
(Option buttons)
  • Record History:
Should SIP history be recorded
Example:
Select 'Yes' to record SIP history. Example history information
 * SIP Call
 1. TxReqRel        INVITE / 102 INVITE
 2. Rx              SIP/2.0 / 102 INVITE /100 Trying
 3. CancelDestroy
 4. Rx              SIP/2.0 / 102 INVITE /180 Ringing
 5. CancelDestroy
 6. Rx              SIP/2.0 / 102 INVITE /200 OK
 7. CancelDestroy
 8. Unhold          SIP/2.0
 9. TxReq           ACK / 102 ACK
 10. TxReqRel        INVITE / 103 INVITE
 11. Rx              SIP/2.0 / 103 INVITE /200 OK
 12. CancelDestroy
 13. Unhold          SIP/2.0
 14. TxReq           ACK / 103 ACK
(Option buttons)
  • Dump History:
Dump SIP history at end of SIP dialogue. SIP history is output to the DEBUG logging channel.
(ex. Yes, No, N/A)
(Option buttons)

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])

Billing

Billing
  • 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.
- billing: Mark the entry for billing
- 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)
  • Disallow:
Set the codecs extension is now allowed to use
(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)

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.

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.

WOOMERA

WOOMERA is the protocol which is used by Sangoma cards.

General Configuration
  • WOOMERA Server:
This is the remote IP address of the Woomera Server
(ex. 192.168.0.13)
([0-9])
  • WOOMERA Server Port:
This is the remote port number of the Woomera Server
(ex. 42420)
([0-9])
  • WOOMERA Audio Server:
IP address of the remote SMG Audio Server
(ex. 192.168.0.13)
([0-9])
  • Default Context:
Incoming Context for Woomera Channels. All incoming calls will be forwared to defined context in extensions.conf
(ex. dids)
([a-z][0-9])
  • Debug:
Debug flag is used to enable/disable the woomera channel debugging level.
(ex. 2)
([0-9])
  • Enable Incoming DTMF:
This option will enable Rx DTMF detection on each channel. The Tx DTMF is automatically enabled on SMG
(ex. Yes, No, N/A)
(Option buttons)
  • Enable JitterBuffer:
Enable Jitterbuffer for this profile
(ex. Yes, No, N/A)
(Option buttons)
  • Enable Progress Messaging:
Enable Asterisk Progress Messaging used for Asterisk Early Audio with ZAP and SIP
(ex. Yes, No, N/A)
(Option buttons)
  • Coding:
Coding is necessary to identify ulaw/alaw
(ex. ulaw)
(Select box)
  • RX Gain:
Set RX gain
(ex. 2)
([0.0 - x.0])
  • TX Gain:
Set TX gain
(ex. 2)
([0.0 - x.0])
  • Language:
Set preferred language
(ex. en)
([a-z])

Providers

PBXware MT 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'.

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


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


PSTN

From here you can select two PSTN types of providers: ZAPTEL (analog and PRI), MISDN (BRI).


ZAPTEL

PSTN
  • Type:
Service type
(ex. pstn/voip)
(Select box)
  • Protocol:
Protocol provider uses
(zaptel/capi)
(Select box)
  • Provider:
Provider name
(ex. BT)
([a-z][0-9])
  • Country:
Destination of the trunk connection
(ex. USA)
(Select box)
  • E164 Accepted:
Does the Provider 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)
  • National Dialing Code:
National dialing code at the Provider destination
(ex. For USA, 1, United Kingdom and Germany 0)
([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. NOTE: Before setting this option to 'Yes', go to 'Settings: Servers' and enable this option as well.
(ex. John dials 121255510205. With this option enabled)
([0-9])
  • International Dialing Code:
International dialing code at the Provider destination
(ex. For USA, 011, United Kingdom and Germany, 00)
([0-9])
  • Local Area Code:
Add local area code to dialed number, if required by the service provider. (By default, the local area code is stripped when dialing)
(ex. User dials 55510205, the local area code is 212. If the call goes through this trunk PBXware MT will dial 21210205)
([0-9])
  • Write dialing code:
Should National and International prefix be written into configuration files
(ex. Enable this option if required by the provider)
([0-9])
  • Zaptel Devices:
Select which zaptel device the system is to use
Example:
  • X100P
  • TDM400P
  • TDM10B
  • TDM20B
  • TDM30B
  • TDM40B
  • TDM11B
  • TDM12B
  • TDM13B
  • TDM21B
  • TDM22B
  • TDM23B
  • T100P
  • E100P
  • TE110P
  • TE410P
  • TE405P
  • hfcISDN
  • quadBRI
  • A101u
  • A102u
  • A104u
(Option buttons)


tip.png Tip

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


MISDN

MISDN
  • Country:
Country which will be shown as a default when adding a trunk with the current provider.
(ex. United Kingdom)
(Select box)
  • National dialing code:
Prefix when dialing national numbers.
(ex. 0)
([0-9])
  • International dialing code:
Prefix used when dialing international numbers.
(ex. 00)
([0-9])
  • E164 Accepted:
Does the provider used 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:
Pass the digits dialed without any conversion (E164, National, Area code). NOTE: When active, 'Leave National Code and 'Local Area Code' will be disabled
(ex. If this option is disabled, PBXware MT will convert all dialed numbers to E164 format (COUNTRY_CODE + AREA_CODE + DIALED_NUMBER) and then make a call to the converted number. If this option is enabled, PBXware MT will call directly DIALED_NUMBER without making any number conversions)
(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. 035123456 will not be stripped of 0)
(Option buttons)
  • Local Area Code:
Add the local area code to the dialed number, if required by the service provider. (By default, the local area code is stripped when dialing)
(ex. User dials 55510205, local area code is 212. If call goes through this trunk, PBXware MT will dial 21210205)
([0-9])
  • Incoming Limit:
Number of simultaneous incoming calls the trunk can handle
(ex. 4 equals four simultaneous incoming calls. Any additional calls will get the busy sound)
([0-9])
  • Outgoing Limit:
Number of simultaneous outgoing calls the trunk can handle
(ex. 4 equals four simultaneous outgoing calls. Any additional calls attempting to use this trunk will be rejected or will be redirected to other trunks 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)
  • mISDN Devices:
Select the mISDN device that the system is to use
Example:
  • B410P
  • USB-ISDN
(Option buttons)


VOIP

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


SIP

SIP
  • 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 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 the E164 format
(ex. Enabling this option will reformat any dialed number into 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
(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: Servers' and enable this options 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 the service provider. (By default, the local area code is stripped when dialing)
(ex. User dials 55510205, local area code is 212. If the call goes through this provider, PBXware MT will dial 21210205)
([0-9])
  • Canreinvite:
Should you allow RTP voice traffic to bypass Asterisk. 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)
(Options buttons / Select box)
  • Default IP:
IP address to be used until registration
(ex. 192.168.1.1)
(IP Address)
  • Incoming Limit:
Number of simultaneous incoming calls the Provider can handle
(ex. 4 equals 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 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)
  • 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 that the system sends the calls to
(ex. 192.1168.1.1)
(IP Address)
  • Auth:
Global credentials for outbound calls, i.e. when a proxy challenges your PBXware MT for authentication these credentials override any credentials in peer/register definition if realm is matched.
(ex. john:dfgERG@pbxware)
([a-z][0-9])
  • Register:
Method for registering to 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])
  • Insecure:
Which of the selected options are not used for authentication.
(ex. very - Do not require authentication of incoming INVITEs)
(Select box)
  • Disallow:
This field is very unique. In order to work properly this setting is automatically set to 'Disallow All' and cannot be modified.
(ex. all)
(Display)
  • Allow:
Codecs that are allowed in 'Settings: Server' will be enabled for selection.
(Check boxes)
  • Auto-Framing
Automatically adjust RTP packet size with the registered device.
(ex. Yes)
(Option buttons)


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


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 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 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
(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: Servers' 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 the service provider. (By default, the local area code is stripped when dialing)
(ex. User dials 55510205, the local area code is 212. If the call goes through this provider, PBXware MT will dial 21210205)
([0-9])
  • Canreinvite:
Should you allow RTP voice traffic to bypass Asterisk. 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)
(Options buttons / Select box)
  • Default IP:
IP address to be used until registration
(ex. 192.168.1.1)
(IP Address)
  • Incoming Limit:
Number of simultaneous incoming calls Provider can handle
(ex. 4 equals four simultaneous incoming calls. Any additional calls will get the busy sound)
([0-9])
  • Outgoing Limit:
Number of simultaneous outgoing calls that a Provider can handle
(ex. 4 equals 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)
  • Notransfer:
Disable the 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)


WOOMERA

WOOMERA
  • 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 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
(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: Servers' and enable this options 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 the service provider. (By default, the local area code is stripped when dialing)
(ex. User dials 55510205, the local area code is 212. If the call goes through this provider, PBXware MT will dial 21210205)
([0-9])
  • Incoming Limit:
Number of simultaneous incoming calls that the Provider can handle
(ex. 4 equals 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 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)


E-mail 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 information on newly created extensions.

New Extension Templates








Billing E-mail Templates

This is used to set E-mail template for extensions Soft and Hard billing limits.

Billing 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







Incoming/Outgoing Limit Templates

Set the e-mail template used for sending emails when Fax-to-email is set on a DID.

Incoming/Outgoing Limit Templates







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 of silence in a voice message in seconds
(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, the more background noise is added)
([0-9])
  • Voicemail Delay:
Delay a number of seconds before asking the user for the 'Password'
(ex. If you hear a partial sound file played asking the user for a 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 the user of new voicemail messages.


E-mail 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' in the email header the following line is added '"$FROM": string <pbxe@domain.com>')
([a-z][0-9])
  • Send Attachment:
Send the voice message as an attachment to the user's email.
(ex. Once B gets the new voice message, if this option is set to 'Yes', the message sound file will be attached to the new voicemail notification email).
(Option buttons)
  • Delete After E-mailing:
Delete a voice message after sending it as an attachment to a user email.
(ex. Once B gets the new voice message, if this option is set to 'Yes', the message will be deleted from the voice box after it has been emailed to B).
(Option buttons)
  • Skip[PBX]: in subject:
Should 'PBX' be skipped from the voicemail title
(ex. If set to 'No' - 'Subject: [PBX']: New message M in mailbox B' will be displayed in email subject line)
(Option buttons)
  • "From:" string:
Override a portion of the From: line in the voicemail notification email
(ex. If this option is set to 'MARS', then the email will have one additional line in its header 'From: MARS')
([a-z][0-9])
  • E-mail Subject:
Customize the voicemail notification email subject
(ex. Use custom variables '${VM_MSGNUM}' for message number and '${VM_MAILBOX}' to create a custom email subject. 'PBXware MT: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}' for example. Email subject would look like this 'PBXware MT: New message 3 in mailbox 1002')
([a-z][0-9])
  • E-mail body:
Customize voicemail notification email body
Example:
Use custom variables '${VM_NAME}' for User name, '${VM_MAILBOX}' for mailbox number, ${VM_DATE} for voicemail date and ${VM_DUR} for voicemail duration to create custom email body. For example:
Dear, ${VM_NAME}\n\nYou have a new voicemail message on:\n\nmailbox: ${VM_MAILBOX}\n\nleft at: ${VM_DATE}\n\n${VM_DUR} long.
([a-z][0-9])
  • Charset:
Select which charset to use when sending mails.
(ex. ISO-8859-1)
  • Mail command:
Overrides the default mailer command
(ex. Default value '/usr/sbin/sendmail -t')

([a-z][0-9])

Run Application

Run custom applications on certain voicemail actions


Ran Application
  • On Voicemail:
Run a custom application when a new voicemail is received
Example:
Set this field to
 '/usr/bin/myapp' 

to execute

 'myapp' 

application when a new voicemail arrives

(([a-z][0-9])
  • 'On Password Change:
Run custom application when a voicemail password is changed
Example:
Set this field to
 '/usr/bin/myapp' 

to execute the

 'myapp' 

application when the voicemail password changes)

([a-z][0-9])


Main Voicemail Menu

Edit settings for the 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 the 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 has been recorded.
(ex. If this option is set to 'Yes', when checking voicemail, the '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 voice message 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 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 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 played (Once '0' is pressed, 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 context to send voicemail from
(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
Example:
Set this field to
 'fromvm' 

for example

([a-z][0-9])
  • Callback context:
Context to all call back
Example:
Set this field to
 'tomv' 

If not listed, calling the sender back will not be permitted

([a-z][0-9])


Edit Timezone

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


Add/Edit Timezone
Add/Edit Timezone
  • Name:
Unique timezone name
(ex. Name provided here will be visible when setting 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 closes timezone to Zenica here (e.g. 'Europe/Sarajevo'))
(Select box)
  • Time format:
Set the appropriate time format
Example:
Depending on 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 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])


Mailbox Option

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

Once the user is transferred to the party's voice box the '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 # key or by hanging up
  2. To reach an operator by dialing 0

If 0 is dialed 'Press 1 to accept this recording, otherwise please continue to hold' message will be heard. 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 is played.
  • 1 Read voicemail messages
  • 2 Change folders
    • 0 Mailbox options
    • 1 Record your unavailable message
    • 2 Record your busy message
    • 3 Record your name
    • 4 Record your temporary message (new in Asterisk v1.2)
    • 5 Change your password
    • * Return to the main menu
  • 3 Advanced options (with option to reply; introduced in Asterisk CVS Head April 28, 2004 with 'enhanced voicemail')
    • 1 Reply
    • 2 Call back(1)
    • 3 Envelope
    • 4 Outgoing call(1)
  • 4 Play previous message
  • 5 Repeat current message
  • 6 Play next message
  • 7 Delete current message
  • 8 Forward message to another mailbox
  • 9 Save message in a folder
  • * Help; during msg playback: Rewind
  • # Exit; during msg playback: Skip forward
  • ** Help
  • *# Exit

After recording a message (incoming message, busy/unavailable greeting, or name)

  • 1 Accept
  • 2 Review
  • 3 Re-record
  • 0 Reach operator(1) (not available when recording greetings/name)


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 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 'Import current file' or 'Import All Files' button to update the files. This error usually happens if configuration file is changed through system console (shell).


tip.png Tip

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



g729

g729
  • Codec optimized for:
Select the server architecture.
(ex. Select the type of server processor)
(Select box)
  • Register Utility:
Path to codec's register utility, required for codec install
(ex. Please note that the default path leads to 32-bit software architecture. For 64-bit please visit http://ftp.digium.com/pub, navigate to desired version and paste the new destination into this field).
([a-z] [0-9] [:/_.-])
  • Licence Key:
Codec license key:
(ex. Provide the codec license key here).
([a-z] [0-9] [_.-])
  • Install:
Install button
(ex. After all necessary details are provided with the correct data(valid urls), click this button to install the g729 codec).
(Command button)


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


Settings: On slave tenants


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.


Defaults 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

UAD (User Agent Devices) are various IP phones, soft phones, ATA (Analog Telephone Adaptors) and IAD (Integrated Access Devices) used for system extensions. PBXware MT supports a wide range of UAD using SIP,IAX, MGCP and ZAPTEL 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 MT 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 UAD not listed, clicking on "Add User Agent" allows the addition of a new UAD.


Requirements

Paging

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

GXP 2000
  1. Navigate in your Internet browser to the telephone's 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 the 'Auto-Answer' options
  6. Select 'Yes'
  7. Click on the 'Update' button
  8. Click on the 'Reboot' button


tip.png Tip

Paging tested with Firmware version 1.0.1.12


Snom 190/320
  1. Navigate in your Internet browser to the telephone's 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 a call has been received
  8. Set 'Type of Answering' to suit your needs e.g. 'Handsfree'
  9. Click on the 'Save' button


tip.png 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 complete the 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 Polycom phone ring type, in this case an answer, that means that phone will automatically answer without ringing.

3. Update the modified files to the provisioning server 4. Reload PBXware MT if used as a provisioning server 5. Restart your telephone The bootROM on the telephone 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 sev-eral 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 unex-pected 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.


tip.png 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.


UAD




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 recog-nized 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 whether 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. This is not good.

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. This is useful for after-hours listening throughout the entire office.


tip.png Tip

Paging tested with Firmware version 6.1


Linksys 941
  1. Navigate in your Internet browser to the telephone's IP address
  2. Select 'Admin Login'
  3. Select 'Advanced'
  4. Navigate to the 'User' tab
  5. Set 'Auto Answer Page' to 'Yes'
  6. Set 'Send Audio To Speaker' to 'Yes'
  7. Click on the 'Submit All Changes' button


tip.png Tip

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


Aastra 480i/9133i/9112i
  1. Navigate in your Internet browser to the telephone's IP address
  2. Select 'Preferences' under '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.png Tip

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


Add/Edit

SIP

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


SIP
  • Protocol:
Select protocol UAD(User Agent Device) uses
(ex. If 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 assign 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)
(Select box)
  • Context:
Every system extension belongs to a certain system context. Context may be described as a collection/group of extensions. Default context used by the PBXware MT 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)
  • Internal UAD name for Auto-Provisioning
This is where you define how the system will recognize UAD internally
(ex. aa57i)
(Select box)
  • NAT (Network ADdress Translation):
Set the appropriate Extension - PBXware MT NAT relation
Example:
If Extension 1000 is trying to register with the PBXware MT 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
  • never - Never attempt NAT mode or RFC3581 support
  • route - Assume NAT, don't send rport
(Option buttons)
  • Canreinvite
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 buttons)
  • Qualify
Timing interval in milliseconds at which a 'ping' is sent to UAD/Phone or trunk, in order to find out its status(online/offline).
(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])
  • Ringtime:
UAD/Phone ring time
(ex. Time in seconds 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 detail 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 detail list of all available dial options( default: empty ))
([a-z])
  • Incoming Limit:
Maximum number of incoming calls
(ex. 2)
([0-9])
  • Outgoing Limit:
Maximum number of outgoing calls
(ex. 2)
([0-9])
  • Disallow:
Set the codecs extension is now allowed to use
(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-Framing (RTP Packetization):
If autoframing is turned on, system will choose packetization level based on remote ends preferences.
(ex. Yes)
(Option buttons)
  • Auto Provisioning:
Enable auto provisioning service for this extension
(ex. Connect UAD/Phone to PBXware MT without any hassle by providing UAD/Phone MAC address( and optionally adding Static UAD/Phone IP address and network details))
(Option Buttons)
  • DHCP ( Dynamic Hosts Configuration Protocol ):
Set whether UAD/Phone is on DHCP or Static IP address
(ex. Set DHCP = Yes if UAD/Phone is on dynamic or DHCP = No if UAD/Phone is on static IP address. If on static IP, you will have to provide more network details in the fields bellow).
(Option buttons)
  • Presence
Presence enables UAD to subscribe to hints which are used for BLF.
(ex. Yes, No, N/A)
(Options buttons)
  • 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])


IAX

Click 'Add User Agent' to add a device or click '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)
  • 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 assign 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)
(Select box)
  • Context:
Every system extension belongs to a certain system context. Context may be described as a collection/group of extensions. Default context used by the PBXware MT 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)
  • NAT (Network Address Translation):
Set the appropriate Extension - PBXware MT NAT relation
Example:
If Extension 1000 is trying to register with the PBXware MT 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
  • never - Never attempt NAT mode or RFC3581 support
  • route - Assume NAT, don't send report
(Option buttons)
  • Canreinvite
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)
  • Qualify
Timing interval in milliseconds at which a 'ping' is sent to UAD/Phone or trunk, in order to find out its status(online/offline).
(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])
  • Ringtime:
UAD/Phone ring time
(ex. Time in seconds 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 detail 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 detail list of all available dial options( default: empty ))
([a-z])
  • Incoming Limit:
Maximum number of incoming calls
(ex. 2)
([0-9])
  • Outgoing Limit:
Maximum 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)
  • Auth Method:
Authentication method required by provider
(ex. md5)
([a-z][0-9])
  • Encryption:
Should encryption be used when authenticating with the peer
([a-z][0-9])
  • Disallow:
Set the codecs extension is now allowed to use
(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:
Enable auto provisioning service for this extension
(ex. Connect UAD/Phone to PBXware MT without any hassle by providing UAD/Phone MAC address( and optionally adding Static UAD/Phone IP address and network details))
(Option Buttons)
  • DHCP ( Dynamic Hosts Configuration Protocol ):
Set whether UAD/Phone is on DHCP or Static IP address
(ex. Set DHCP = Yes if UAD/Phone is on dynamic or DHCP = No if UAD/Phone is on static IP address. If on static IP, you will have to provide more network details in the fields bellow).
(Option buttons)
  • Presence
Presence enables UAD to subscribe to hints which are used for BLF.
(ex. Yes, No, N/A)
(Options buttons)
  • 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])


ZAPTEL

ZAPTEL



















Zapata General
Zapata General
  • Device Name:
Unique device name
(AASTRA 480i)
([a-z][0-9])
  • Channels
Which card channels are used
(ex. 1,4/1-4)
([0-9][,-])
  • Language:
Default language
(ex. us)
(Select box)
  • 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)
  • Signaling:
Signaling method
Example:
Default
  • FXS Loopstart
  • FXS Groundstart
  • FXS Kewlstart
  • FXO Loopstart
  • FXO Groundstart
  • FXO Kewlstart
  • PRI CPE side
  • PRI Network side
  • BRI CPE side
  • BRI Network side
  • BRI CPE PTMP
  • BRI Network PTMP
(Select box)
  • 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])


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 the dial plan used by some switches
Example:
  • Unknown
  • Private ISDN
  • Local ISDN
  • National ISDN
  • International ISDN
(Select box)
  • PRI Local Dial Plan:
Set the numbering dial plan for destinations called locally
Example:
  • Unknown
  • Private ISDN
  • Local ISDN
  • National ISDN
  • International ISDN
(Select box)
  • PRI Trust CID:
Trust the provided caller id information
(ex. Yes, No, N/A)
(Option buttons)
  • PRI Indication:
How to report 'busy' and 'congestion' on a PRI
Example:
  • 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 the switch, select the network specific facility
Example:
  • 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])
  • 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)
(Options buttons)
  • Hide Caller ID:
Whether or not to hide outgoing caller ID
(ex. Yes, No, N/A)
(Options 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)
(Options buttons)
  • CID Signaling:
Set the type of caller ID signaling
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)
(Options buttons)
  • Send CallerID After:
Some countries, like the 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 Canceller
Echo Canceller
  • Echo Cancel:
Enable echo cancellation
(ex. Yes, No, N/A)
(Option Button)
  • 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)


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, the call can be put on hold and one is able to make another call
(ex. Yes, No, N/A)
(Option buttons)
  • Transfer:
Support call transfer and also enables 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)
  • PulseDial:
Use pulse dial instead of DTMF. Used by FXO (FXS signaling) 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 to which the Call Group 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 group's extension as allowed to pick up.
(ex. Similar to 'Context' grouping, only this option sets the Call Groups extension as allowed to pick up 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 Zapata 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 sync source
  • 1 - use as primary sync source
  • 2 - set as secondary and so forth
([a-z])
  • Line build out:
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 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)
  • Dynamic span address:
Driver specific address (like a MAC for eth).
  • Dynamic span channels:
Number of channels
  • 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.


FXO Channels
FXO Channels
  • FXO Loopstart:
Channel(s) are signaled using FXO Loopstart protocol
  • FXO Groundstart:
Channel(s) are signaled using FXO Groundstart protocol
  • FXO Kewlstart:
Channel(s) are signaled using FXO Kewlstart protocol


The 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)


FXS Channels
FXS Channels
  • FXS Loopstart:
Channel(s) are signaled using FXS Loopstart protocol
  • FXS Groundstart:
Channel(s) are signaled using FXS Groundstart protocol
  • FXS Kewlstart:
Channel(s) are signaled using FXS Kewlstart protocol


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)


PRI Channels
PRI Channels
  • D-Channel(s):
For example, every ISDN BRI card has 1 D- (control) channel
(ex. 1)
([0-9])
  • B-Channels(s):
For example, every ISDN BRI card has 2 B- (data) channels
(ex. 2)
([0-9])


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


WOOMERA

WOOMERA
  • Protocol:
Select protocol UAD(User Agent Device) uses
(ex. If UAD/Phone uses MGCP protocol, select 'MGCP' here)
(Select box)
  • Device Name:
Unique device name
(ex. AASTRA 480i)
([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)
  • User Agent Auto Provisioning Template:
This option is used for providing additional config parameters for WOOMERA configuration files. The values provided here will be written into these configuration files.
([a-z][0-9])


Access Codes

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

Access codes

Voicemail

  • Voicemail:
Voice inbox access code. This number is dialed to access extension voice inbox (extension PIN required)
(ex. From extension 1000 dial '*123' to access extension 1000 voice inbox. When asked for PIN, provide 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 dialling the last Extension that was calling you
(ex. If Extension '1000' was the last extension calling 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])
  • Instant Recording:
Access code for instant call recording
(ex. During active conversation dial '*159' to start recording conversation)
([0-9])
  • Pause/UnPause Recording:
If making an outbound call or being a logged in as an Agent, you have the ability to pause or unpause call recording using this access code
(ex. Whether you need to hide sensitive information like credit card numbers or anything else not being recorded, you can use this access code to pause or unpause the call recording in progress)
([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])
  • Monitor Conference:
This is the access code for Conference monitoring and its usage is described in Enhanced Services of the extensions
(ex. 500+CONF_NUM)
([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])

PBXware 3.8 brings in new feature in form of Two Way Speakerphone Paging. In order to start two way communication with preferred extension on the system dial *400 + * + $EXTENSION. For example, if you would like to initiate two way speakerphone paging with extension 1038 you will dial: *400*1038.

Features

  • Speed Dial:
Speed Dial enables faster dialing by entering an 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 networks
(ex. 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 to the 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).
([1-9])

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" Enhanced Service.
Dial *78 to enable "Do Not Disturb" and use set of rules defined for this service in extensions Enhanced Services.
(*[0-9])
  • Disable Do Not Disturb:
Access Code for disabling "Do Not Disturb" Enhanced Service.
Dial *79 to disable "Do Not Disturb" and use set of rules defined for this service in extensions Enhanced Services.
(*[0-9])
  • Enable Group Hunt:
Access Code for enabling "Group Hunt" Enhanced Service.
Dial *510 to enable "Group Hunt" and use set of rules defined for this service in extensions Enhanced Services.
(*[0-9])
  • Disable Group Hunt:
Access Code for enabling "Group Hunt" Enhanced Service.
Dial *511 to enable "Group Hunt" and use set of rules defined for this service in extensions Enhanced Services.
(*[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 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). 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])
  • Enhanced Call Park:
Access code for parking calls and sending them to preset Announce Extension which can be set in Settings->Servers
(ex. During active conversation dial '#800'. The call will be parked and Announce Extension will ring for Timeout seconds. After that period call will be directed to Timeout Extension)
([0-9])

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])

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])
  • Queue Interrupt Message:
Record an interrupt message to be played to callers waiting in queue
(ex. When a user on the system dials this access code, system will ask him to enter agent number, PIN, and a queue to which he wants to play the message. When that info is entered user will record a message after which he will press #. System will ask if you want to review your message, record it again or confirm recorded message. When you confirm message it will be played to all callers waiting in given queue).
([0-9])

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])

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])

Numbering Defaults

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

Numbering Defaults




Fetch least unallocated number (default):

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


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: Business, Version: 2.0.0, Release: 2007-06-27 (#1, $Revision: 2135 $), Running: 1.2.13-b20070521)
(Display)
  • Package:
Package information displays a number of system Extensions, Trunks, Conferences, etc.
(ex. Servers: 1, Extensions 768...)
(Display)
  • Enhanced Services:
This line identifies the system's Enhanced Services
(ex. Last Caller, Group Hunt, Call Forwarding...)
(Display)
  • License Details:
This line displays license details.
NOTE: If 'Branding' is enabled, only the license number is visible
(ex. License No: 7E5CF50C)
(Display)


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