Mode

Network port

FXS Ports

FXO Ports

USB

SD

UPS

UC300-A11EM1

1 x 10/100M Ethernet

1

1

1

1

NO

UC300-A14EM1

1 x 10/100M Ethernet

1

4

1

1

NO

UC300-A02EM1

1 x 10/100M Ethernet

0

2

1

1

NO

UC300-A11EM1

2 x 10/100M Ethernet

1

1

1

1

YES

UC300-A14EM2

2 x 10/100M Ethernet

1

4

1

1

YES

UC300-A02EM2

2 x 10/100M Ethernet

0

2

1

1

YES

UC500-A22EM2

2 x 10/100M Ethernet

2

2

1

1

YES

UC500-A44EM2

2 x 10/100M Ethernet

4

4

1

1

YES

UC500-A08EM2

2 x 10/100M Ethernet

0

8

1

1

YES

Item 

Description

(1) System button area

Contains buttons such as Change Password, Reboot, Logout, Skins, Language, etc., and the event notification bar, displays the current login user.

(2) Menu bar

Displays submenus for your selection when the mouse pointer is moved onto a menu. The selection result is displayed in the configuration area.

(3) Configuration area

View or modify configurations.

Item

Description

Basic Settings

Host

Server Name, for example: pbx.subdomain.com

Work Mode

Optional work modes: Single/Double

Gateway

IP Address of the Port of Connection (Default Gateway)

Primary DNS

IP Address of the Primary Domain Name Server (DNS)

Secondary DNS

IP Address of the Secondary or Alternative Domain Name Server (DNS)

IP Configuration

Type

The type of IP address that the Interface has, which could be STATIC when the IP address is fixed or DHCP when the IP address is obtained automatically from a DHCP server.

IP Address

IP Address assigned to the Interface

Mask

The Network Mask assigned to the Interface

MAC

Physical Address of the network Interface

Status

Shows the physical status of the Interface, if it’s connected or not

Default Route

Mainly used in Double work mode to determine the default exit for network traffic

IP Address 2

The second IP assigned to the Interface

Mask 2

The network mask for the second IP

Item

Description

Enable

It indicates if the DHCP service is enabled or disabled

Interface

Specify the start IP of the port (Network interface configuration must be static).

Start range of IPs

This will be the beginning of the IP range that the server will provide.

End ranges of IPs

This will be the ending of the IP range that the server will provide.

Address Lease time

Duration for DHCP server to lease an address to a new device. When the lease expires, the DHCP server might assign the IP address to a different device. Default value is 7200 seconds.

DNS 1

This address is the Primary DNS that the server will provide.

DNS 2

This address is the Secondary DNS that the server will provide.

WINS

It is the IP of the WINS Server that will be given to Windows machines.

Gateway

This is the address the server will provide as Gateway.

TFTP

Enter the TFTP server address if required which may be used to auto provision your IP phones.

Status

Display current DHCP status.

Options

Description

HA-Mode

Peer-Peer hot standby mode

Mode

The default is slave mode. The device that turns on the hot standby firstly is the master server.

Sync NIC

The network adapter which is used to heartbeat and synchronous data.

Local Hostname

Hostname of the local host

Peer Hostname

Hostname of the peer host

Peer IP Address

IP address of the peer host

Encryption key

A phrase or password to use for encryption. It has to match on both nodes.

Virtual IP Address

Enter an unused IP address. The extensions would communicate with the server via the virtual IP address. The two PBX in the hot standby mode should configure the same Virtual IP address.

Advert Time

It sets the interval at which Heartbeat keep-alive packets are sent. The default is 2s, the default dead time is 3 * advert time.

Parameter

Importance

Restriction

Default Value

Name

Required

·         The following characters are NOT allowed:

& " ' \ < > ` |

·         The maximum length is 127.

N/A

Description

Optional system services and custom services

·         The following characters are NOT allowed:

& " ' \ < > ` |

·         The maximum length is 511.

N/A

Service

Optional system services and custom services

Permitted value:

HTTP HTTPS SSH AMI SIP-UDP SIP-TCP SIP-TCP SIP-TLS SIP-RTP WEBRTC IAX2 LDAP MYSQL Custom

Custom

Action

Required

Permitted value:

Accept Reject Drop

Accept

Protocol

Required

Permitted value:

udp tcp both

udp

MAC Address

Optional system services and custom services

MAC address format required.

N/A

Type

Required

Permitted value:

IP Domain

IP

Source IP Address/Subnet Mask

Required if Type is IP

IP format required.

N/A

Domain

Required if Type is domain

Domain format required.

N/A

Port

Required

The valid port range is 0-65535.

N/A

Type

Option

Description

CDR Permission

Download CDR

Allow the extension to download the CDR associated with this extension in the Me module

Delete CDR

Allows the extension to delete the CDR associated with this extension in the Me module

Recording Permission

Play Recordings

Allows extensions to play recordings in the Me module

Download Recordings

Allows extensions to download recordings in the Me module

Delete Recordings

Allow extensions to delete recordings in the Me module

Allowed check and Download

Allows extensions to view downloads from other extensions in the "Downloads" section of the Me Bar

Item

Definition

Enable

Decide whether to turn on SMTP service

SMTP Server

SMTP server type. Multiple server types are built in, associated with Domain, or can be customized by selecting "other"

Domain

SMTP server address. It is automatically filled according to the SMTP server. When the SMTP Server selects "other", it needs to be filled manually.

Port

Port to establish the connection with SMTP Server. Common ports are 25, 465 (SSL), 587 (SSL)

Username

Username of email account from SMTP Server.

Password

Password of email account from SMTP Server

TLS Enable

To enable certificates of TLS (Transport Layer Security). Generally, this check is required when using ports that require SSL encryption, such as 465/587. If checked when using a port that does not require encryption, it will cause the send to fail

Options

Definition

Enable

Yes/No. Indicates that the cloud management function is enable/disable

Account

An account or email registered on the cloud management platform.    

Password

Password for the account registered on the cloud management platform.

Server

Three servers are currently supported, including American, China and Europe.

Connect Status

Whether or not you are currently connected to a cloud management platform.

Item

Description

Basic

User Extension

The extension number to dial to reach this user.

Display Name

The CallerID name for calls from this user will be set to this name. only enter the name, NOT the number.

Registration Password

Password configured for the extension to register.

Email Address

The email can be used to email notification to the extension user.

Mobile Number

The extension contacts phone number.

User Password

Password configured for the extension to login web.

Advanced

Account Code

Account code for the device

Max Contacts

Maximum number of endpoints that can associate with this device

Web Phone

Enable web phone will let user make and receive calls without installing any plugin in web browser.

Dtmfmode

The DTMF signaling mode used by this device, usually rfc2833 for most phone.

Audio Codecs

Codecs supported by the device, you can choose the codecs which you want.

Video Codecs

Video codecs supported by the device

Ring Timeout

Select the time in seconds.

Transport

This sets the allowed transport settings for this device and the default (Primary) transport for outgoing. The default transport is only used for outbound message until a registration takes place. During the peer registration the transport type may change to another supported type if the peer requests so. In most common cases, this does not have to be changed as most devices register in conjunction with the host=dynamic setting. If you are using TCP and/or TLS you need to make sure the general SIP Settings are configured for the system to operate in those modes and for TLS, proper certificates have been generated and configured. If you are using websockets (such as WebRTC) then you must select an option that includes WS.

User Agent

When registering, SIP phones will be sending packets containing the user agent string. If the prefix of the user agent does not match the value defined here, the registration will fail.

Permitted IP/Subnet Mask

Permitted IP/Subnet Mask

Dictation Service

Allow the device to support dictation service.

Dictation Format

The format of dictation.

Language Code

Choose a different language for the user if he/she is not a native speaker than default system voice prompts.

CID Num Alias

The CID Number to use for internal calls, if different from the extension number. This is used to masquerade as a different user.

SIP Alias

If you want to support direct sip dialing of users internally or through anonymous sip calls, you can supply a friendly name that can be used in addition to the users extension to call them.

Features

Outbound CID

Override the callerid when dialing out a trunk. Any setting here will override the common outbound callerid set in the trunk admin.

Format: “caller name” <#######>

Leave this filed blank to disable the outbound callerid feature for this user.

Asterisk Dial Options

Cryptic Asterisk Dial Options, check to customize for this extension or un-check to use system defaults set in Advanced Options. These will not apply to trunk options which are configured with the trunk.

Ring Time

Number of seconds to ring prior to going to voicemail. Default will use the value set in Advanced Settings. If no voicemail is configured this will be ignored.

Allow Being Monitored

Check this option to allow this user to be monitored.

Monitor Mode

Decide how you will monitor another extension.

Call Forward Ring Time

Number of seconds to ring during a Call Forward Busy or Call Forward Unavailable call prior to continuing to voicemail or specified destination. Setting to Always will not return, it will just continue to ring. Default will use the current Ring Time. If voicemail is disabled and there is not destination specified, it will be forced into Always mode.

Outbound Concurrency Limit

Maximum number of outbound simultaneous calls that an extension can make. This is also very useful as a Security Protection against a system that has been compromised. It will limit the number of simultaneous calls that can be made on the compromised extension.

Call Waiting

Set the initial/current Call Waiting state for this user’s extension

Internal Auto Answer

When set to Intercom, calls to this extension/user from other internal users act as if they were intercom calls meaning they will be auto-answered if the endpoint supports this feature and the system is configured to operate in this mode. All the normal white list and black list settings will be honored if they are set. External calls will still ring as normal, as will certain other circumstances such as blind transfers and when a Follow Me is configured and enabled. If Disabled, the phone rings as a normal phone.

Call Screening

Call Screening requires external callers to say their name, which will be played back to the user and allow the user to accept or reject the call. Screening with memory only verifies a caller for their callerid once. Screening without memory always required a caller to say their name. Either mode will always announce the caller based on the last introduction saved with that callerID. If any user on the system uses the memory option, when that user is called, the caller will be required to re-introduce themselves and all users on the system will have that new introduction associated with the caller’s CallerID.

Pinless Dialing

Enabling Pinless Dialing will allow this extension to bypass any pin codes normally required on outbound calls.

Emergency CID

This callerid will always be set when dialing out an Outbound Route flagged ad Emergency. The Emergency CID overrides all other CallerID settings.

Queue State Detection

If this extension is part of a Queue will attempt to use the user’s extension state or device state information when determining if this queue member should be called. In some uncommon situations such as a Follow-Me with no physical device, or some virtual extension scenarios, the state information will indicate that this member is not available when they are. Setting this to ‘Ignore-State’ will make the Queue ignore all state information thus always trying to contact this member. Certain side effects can occur when this route is taken due to the nature of how Queues handle Local channels, such as subsequent transfers will continue to show the member as busy until the original call is terminated. In most cases, this SHOULD BE set to ‘Use State’.

Recording

On Demand Recording

Enable or disable the ability to do on demand (one-touch) recording. The overall calling policy rules still apply and if calls are already being recorded they cannot be paused.

Record Priority Policy

Call recording policy priority relative to other extensions when there is a conflict between an extension wanting recording and the other not wanting it. The higher of the two determines the policy, on a tie the global policy (caller or callee) determines the policy.

Voicemail

Status

Enable or disable the voicemail function.

Voicemail Password

This is the password used to access the Voicemail system.

This password can only contain numbers.

A user can change the password you enter here after logging into the Voicemail system (*98) with a phone.

Pager Email Address

Page/mobile email address that short Voicemail notifications are sent to.

Email Attachment

Option to attach Voicemail to email.

Play CID

Read back caller’s telephone number prior to playing the incoming message, and just after announcing the date and time the message was left.

Play Envelope

Envelope controls whether or not the Voicemail system will play the message envelope (date/time) before playing the voicemail message. This setting does not affect the operation of the envelope option in the advanced voicemail menu.

Delete Voicemail

If set to “yes” the message will be delete from the voicemailbox (after having been emailed). Provides functionality that allows a user to receive their voicemail via email alone, rather than extension handset. CAUTION: must have attach voicemail to email set to yes otherwise your messages will be lost forever.

Send Voicemail

If set to 'yes', the voicemail will be sent by email.

VM Options

Separate options with pipe( | )

Ie: review=yes|maxmessage=60

VM Context

This is the voicemail context which is normally set to default. Do not change unless you understand the implications.

Routing

VmX Locater™

Enable/ disable the VmX locater feature for this user. When enabled all settings are controlled by the user in the user portal (ARI). Disabling will not delete any existing user settings but will disable access to the feature.

Use When

Menu options below are available during your personal voicemail greeting playback.

Check both to use at all times.

Voicemail Instructions

Uncheck to play a deep after your personal voicemail greeting.

Press 0

Pressing 0 during your personal voicemail greeting goes to the operator. Uncheck to enter another destination here. This feature can be used while still disabling VmX to allow an alternative operator extension without requiring the VmX feature for the user.

Press 1

The remaining options can have internal extensions, ringgroups, queues and external numbers that may be rung. It is often used to include your cell phone. You should run a test to make sure that the number is functional any time a change is made so you don’t leave a caller stranded or receiving invalid number messages.

Press 2

Use any extensions, ringgroups, queues or external numbers.

Remember to re-record your personal voicemail greeting and include instructions. Run a test to make sure that the number is functional.

No Answer

Optional destination call is routed to when the call is not answered on an otherwise idle phone. If the phone is use and the call is simply ignored, then the busy destination will be used.

CID Prefix

Optional CID prefix to add before sending to this no answer destination.

Busy

Optional destination the call is route to when the phone is busy or the call is rejected the user. This destination is also used on an unanswered call if the phone is in use and the user choose not pickup the second call.

CID Prefix

Optional CID prefix to add before sending to this busy destination.

Not Reachable

Optional destination the call is routed when the phone is office, such as a softphone currently off or a phone unplugged.

CID Prefix

Optional CID prefix to add before sending to this not reachable destination.

Item

Definition

Basic

Ring-Group Number

The number users will dial to ring extensions in this ring group

Group Description

Provide a descriptive title for this Ring Group.

Ring Strategy

Ringall : Ring all available channels until one answers (default)

Hunt: Take turns ringing each available extension

Memoryhunt: Ring first extension in the list, then ring the 1 st and 2 nd extension, then ring 1 st and 2 nd and 3 rd extension in the list…etc.

*-prim: there mode act as described above. However, if the primary extension (first in list) is occupied, the other extensions will not be rung. If the primary is CF unconditional, then all will be rung

First available: ring only the first available channel

Firstnotonphone: ring only the first channel which is not offhook-ignored CW.

Ring Time (max 300 sec)

Time in seconds that the phones will ring. For all hunt style ring strategies, this is the time for each iteration of phone(s) that are rung.

Extension List

List extensions to ring, one per line, or use the Extension Quick Pick below to insert them here.

You can include an extension on a remote system, or an external number by suffixing a number with a ‘#’. Ex:2448089# would dial 2448089 on the appropriate trunk (see outbound routing)

Extension without a ‘#’ will not ring a user’s Follow-Me. To dial Follow-Me, Queues and other numbers that are not extensions, put a ‘#’ at the end.

Destination if no answer

If there is no answer, the call will be sent to the destination.

Advanced

Announcement

Message to be played to the caller before dialing this group.

To add additional recordings please use the “System Recordings” MENU to the left.

Play Music On Hold

If you select a music on hold class to play, instead of ‘Ring’, they will hear that instead of Ringing while they waiting for someone to pick up.

CID Name Prefix

You can optionally prefix the callerid name when ringing extensions in this group, ie: If you prefix with “Sales:”, a call from John Doe would display as “Sales: John Doe” on the extensions that ring.

Alert Info

ALERT_INFO can be used for distinctive ring with SIP devices.

Ignore CF Settings

When checked, agents who attempt to Call Forward will be ignored, this applies to CF, CFU and CFB.  Extensions entered with ‘#’ at the end, for example to access the extension’s Follow-Me, might not honor this setting.

Enable Call Pickup

Checking this will allow calls to the ring group to be picked up with the directed call pickup feature using the group number. When not checked, individual extensions that are part of the group can still be picked up by doing a directed call picked to the ringing extension, which works whether or not this is checked.

Skip Busy Agent

When checked, agents who are on an occupied phone will skipped as if the line were returning busy. This means that call waiting or multi-line phones will not be presented with the call and in the various hunt style ring strategies, the next agent will be attempted.

Confirm Calls

Enable this if you’re calling external numbers that need confirmation-eg, a mobile phone may go to voicemail which will pick up the call. Enabling this requires the remote side push 1 on their phone before the call is put through. This feature only works with the ringall ring strategy.

Remote Announce

Message to be played to the person RECEIVING the call, if ‘Confirm Calls’ is enabled.

To add additional recordings use the “System Recordings” MENU to the left

Too-Late Announce

Message to be played to the person RECEIVING the call, if the call has already been accepted before they push 1.

To add additional recordings use the “System Recordings” MENU to the left

Mode

Default: Transmits the Callers CID if allowed by the trunk.

Fixed CID Value: Always transmit the Fixed CID Value below.

Outside Calls Fixed CID Value: Transmit the Fixed CID Value below on calls will continue to operate in default mode.

Use Dialed Number: Transmit the number that was dialed as the CID for calls coming from outside. Internal extension to extension calls will continue to operate in default mode. There must be a DID on the inbound route for this. This will be BLOCKED on trunks that block foreign Caller ID

Force Dialed Number: Transmit the number that was dialed as the CID for calls coming from outside. Internal extension to extension calls will be continue to operate in default mode. There must be a DID on the inbound route for this. This WILL be transmitted on trunks that block foreign CallerID

Fixed CID Value

Fixed value to replace the CID with used with some of the modes above. Should be in a format of digits only with an option of E164 format using a leading “+”.

Record Calls

You can always record calls that come into ring group, never record them, or allow the extension that answers to do on-demand recording. If recording is denied then one-touch on demand recording will be blocked.

Item

Definition

Basic

Extension

Edited extension

Disable

By default (not checked) any call to this extension will go to this Follow-Me instead, including directory calls by name from IVRs. If checked, calls will go only to the extension.

However, destinations that specify FollowMe will come here.

Checking this box is often used in conjunction with VmX Locater, where you want a call to ring the extension, and then only if the caller chooses to find you do you want it to come here.

Initial Ring Time

This is the number of seconds to ring the primary extension prior to proceeding to the follow-me list. The extension can also be included in the follow-me list. A 0 setting will bypass this

Ring Strategy

Ringallv2: ring Extension for duration set in Initial Ring Time, and then, while continuing call to extension, ring Follow-Me List for duration set in Ring Time.

Ringall: ring Extension for duration set in Initial Ring Time, and then, terminate call to extension, ring Follow-Me List for duration set in Ring Time.

Hunt: take turns ringing each available extension

Memoryhunt: ring first extension in the list, then ring the 1st and 2nd extension, then ring 1st 2nd and 3rd extension in the list…. etc.

*-prim: these mode act as described above. However, if the primary extension (first in the list) is occupied, the other extensions will not be rung. If the primary is DND, it won’t be rung. If the primary is CF unconditional, then all will be rung

Firstavailable: ring only the first available channel

Firstavailable: ring only the first channel which is not off hook-ignore CW

Ring Time (max 60 sec)

Time in second that the phones will ring. For all hunt style ring strategies, this is the time for each iteration of phone(s) that are rung

Destination if no answer

Choose a destination when there is no answer.

Follow-Me List

List extensions to ring, one per line, or use the Extension Quick Pick below.

You can include an extension on a remote system, or an external number by suffixing a number with a pound (#). Ex:2448089# would dial 2448089 on the appropriate trunk （see Outbound Routing）.

Announcement

Message to be played to the caller before dialing this group.

To add additional recordings please use the “System Recordings” MENU to the left.

Play Music On Hold

If you select a Music on Hold class to play, instead of ‘Ring’, they will hear that instead of Ringing while they are waiting for someone to pick up.

CID Name Prefix

You can optionally prefix the Caller ID name when ringing extensions in this group. Ie: if you prefix with “Sales:”, a call from John Doe would display as “Sales: John Doe” on the extensions that ring

Alert Info

You can optionally include an Alert Info which can create distinctive ring on SIP phones.

Advanced

Confirm Calls

Enable this if you’re calling external numbers that need confirmation, eg, a mobile phone may go to voicemail which pick up the call. Enabling this require the remote side push 1 on their phone before the calls is put through. This feature only works with the ringall/ringall-prim ring strategy.

Remote Announce

Message to be played to the person RECEIVING the call, if ‘Confirm Calls” is enabled.

To add additional recordings use the ‘System Recordings” MENU to the left

Too-Late Announce

Message to be played to the person RECEIVING the call, if the call has already been accepted before they push 1.

To add additional recordings use the ‘System Recordings” MENU to the left

Mode

Default: Transmits the Caller CID if allowed by the trunk.

Fixed CID Value: Always transmit the Fixed CID Value below.

Outside Calls Fixed CID Value: Transmit the Fixed CID Value below on calls will continue to operate in default mode.

Use Dialed Number: Transmit the number that was dialed as the CID for calls coming from outside. Internal extension to extension calls will continue to operate in default mode. There must be a DID on the inbound route for this. This will be BLOCKED on trunks that block foreign Caller ID

Force Dialed Number: Transmit the number that was dialed as the CID for calls coming from outside. Internal extension to extension calls will be continue to operate in default mode. There must be a DID on the inbound route for this. This WILL be transmitted on trunks that block foreign CallerID

Fixed CID Value

Fixed value to replace the CID with used with some of the modes above. Should be in a format of digits only with an option of E164 format using a leading “+”.

Item

Definition

Basic

Enable Trunk

Check this to disable this trunk in all routes where it is used.

Trunk Mode

Authentication mode of this trunk.

Authentication

Usually, this will be set to "Outbound", which authenticates calls going out, and allows unauthenticated calls in from the other server. If you select "None", all calls from or to the specified SIP Server are unauthenticated.

Trunk Name

Descriptive Name for this trunk.

Host

Host settings for this device, almost always dynamic for endpoint.

Transport

Transports which the device supports.

From user

Rewrite the caller id

From Domain

Example: proxy.provider.domain

Enable NAT

Check this to enable or disable NAT

Codec

Allow specified codecs, the available codecs are on the left options bar and the selected on the right.

Advanced

DTMF Mode

Types of DTMF.

Outbound CallerID

CallerID for calls placed out on this trunk

Format: <#######>. You can also use the format: “hidden” <#######> to hide the CallerID sent out over Digital lines if supported (SIP/IAX).

Maximum Channels

Controls the maximum number of outbound channels (simultaneous calls) that can be used on this trunk. To count inbound calls against this maximum, use the auto-generated context: as the inbound trunk's context. (see extensions_additional.conf) Leave blank to specify no maximum.

Permanent Auth Rejection

Determines whether failed authentication challenges are treated as permanent.

Forbidden Retry Interval

How long to wait before retry when receiving a 403 Forbidden response.

Fatal Retry Interval

How long to wait before retry when receiving a fatal response.

General Retry Interval

The interval between two registered request packets.

Expiration

Expiration time for registrations in seconds.

Max Retries

The times asterisk will attempt to register before give up.

Qualify Frequency

Interval between two qualifies.

Qualify Timeout

Timeout of qualify

Contact User

Contact user to use in request.

AOR Contact

Permanent contacts assigned to AoR.

Support Path

When the button is enabled, registering request of outbound will advertise support for path header.

Support T.38 UDPTL

Allow the device to support T.38 UDPTL

T.38 UDPTL Error Correction

T.38 UDPTL error correction method

T.38 UDPTL NAT

Whether NAT support is enabled on UDPTL sessions

Fax Detect

When a CNG is detected, the session will be sent to the fax extension.

Inband Progress

Determine whether chan_sip indicates ringing using inbound progress.

Direct Media Method

Method for building direct media between endpoints.

Trust Connected Line

Accept Connected Line updates from this endpoint.

Send Connected Line

Send Connected Line updates to this endpoint

Connected Line Method

Method used when updating connected line information.

Direct Media

Determines whether media may flow directly between endpoints.

RTP Symmetric

Enforce that RTP must be symmetric.

Rewrite Contact

Allow contact header to be rewritten

Asterisk Trunk Dial Options

Asterisk Dial command options to be used when calling out this trunk. To override the Advanced Settings default, check the box and then provide the required options for this trunk

Context

(Experts Only) Set the context that calls will originate from. Leaving this as from-internal unless you know what you’re doing.

Continue if Busy

Normally the next trunk is only tried upon a trunk being ‘Congested’ in some form, or unavailable. Checking this box will force a failed call to always continue to the next configured trunk or destination even when the channel reports BUSY or INVALID NUMBER.

Item

Definition

Basic

Enable Trunk

Check this to disable this trunk in all routes where it is used.

Trunk Name

Descriptive Name for this trunk.

Group ID

FXO channels are referenced either by a group number or channel number (which is defined in chan_dahdi.conf). The default setting is g0 (group zero).

Policy

Used to make FXO trunks decisions, help determine the ringing order among multiple members of group

Member of Groups

Adding FXO ports into trunk groups allow automatic selection of the selected idle port for outgoing calls.

Advanced

Outbound CallerID

CallerID for calls placed out on this trunk

Format: <#######>. You can also use the format: “hidden” <#######> to hide the CallerID sent out over Digital lines if supported (SIP/IAX).

CID Options

Determines what CIDs will be allowed out this trunk. IMPORTANT: EMERGENCY

CIDs defined on an extension/device will ALWAYS be used if this trunk is part of an EMERGENCY Route regardless of these settings.

Allow Any CID:  all CIDs including foreign CIDS from forwarded external calls will be transmitted.

Block Foreign CIDs: blocks any CID that is the result of a forwarded call from off the system. CIDs defined for extensions/users are transmitted.

Remove CNAM: this will remove CNAM from any CID sent out this trunk

Force Trunk CID: Always use the CID defined for this trunk except if part of any EMERGENCY Route with an EMERGENCY CID defined for the extension/device. Intra-Company Routes will always transmit an extension’s internal number and name.

Maximum Channels

Controls the maximum number of outbound channels (simultaneous calls) that can be used on this trunk. Inbound calls are not counted against the maximum. Leave blank to specify no maximum.

Asterisk Trunk Dial Options

Asterisk Dial command options to be used when calling out this trunk. To override the Advanced Settings default, check the box and then provide the required options for this trunk

Context

(Experts Only) Set the context that calls will originate from. Leaving this as from-internal unless you know what you’re doing.

Continue if Busy

Normally the next trunk is only tried upon a trunk being ‘Congested’ in some form, or unavailable. Checking this box will force a failed call to always continue to the next configured trunk or destination even when the channel reports BUSY or INVALID NUMBER.

Item

Definition

Basic

Enable Trunk

Check this to disable this trunk in all routes where it is used.

Trunk Mode

Authentication mode of this trunk.

Trunk Name

Descriptive Name for this trunk

Host

Host settings for this device, almost always dynamic for endpoint.

Type

Asterisk connection type. There are three type you can choose, friend, peer and user.usually friend for endpoint.

Trunk

Use IAX2 trunk with this host.

Advanced

Outbound CallerID

CallerID for calls placed out on this trunk

Format: <#######>. You can also use the format: “hidden” <#######> to hide the CallerID sent out over Digital lines if supported (SIP/IAX).

CID Options

Determines what CIDs will be allowed out this trunk. IMPORTANT: EMERGENCY

CIDs defined on an extension/device will ALWAYS be used if this trunk is part of an EMERGENCY Route regardless of these settings.

Allow Any CID:  all CIDs including foreign CIDS from forwarded external calls will be transmitted.

Block Foreign CIDs: blocks any CID that is the result of a forwarded call from off the system. CIDs defined for extensions/users are transmitted.

Remove CNAM: this will remove CNAM from any CID sent out this trunk

Force Trunk CID: Always use the CID defined for this trunk except if part of any EMERGENCY Route with an EMERGENCY CID defined for the extension/device. Intra-Company Routes will always transmit an extension’s internal number and name.

Maximum Channels

Controls the maximum number of outbound channels (simultaneous calls) that can be used on this trunk. To count inbound calls against this maximum, use auto-generated context: from-trunk-[trunkname] as the inbound trunk’s context. (see extesions_additional.conf) Leave blank to specify no maximum.

Outbound Dial Prefix

The outbound dialing prefix is used to prefix a dialing string to all outbound calls placed on this trunk. For example, if this trunk is behind another PBX or is a Centrex line, then you would put 9 here to access an outbound line. Another common use is to prefix calls with 'w' on a POTS line that need time to obtain dial tone to avoid eating digits.

Most users should leave this option blank.

Qualifyfreq OK

Frequency in milliseconds to send qualify messages to the endpoint.

Qualifyfreq Not OK

Frequency in milliseconds to send qualify messages to the endpoint.

Qualify

Setting to yes (equivalent to 2000 msec) will send an OPTIONS packet to the endpoint periodically (default every minute). Used to monitor the health of the endpoint. If delays are longer then the quality time, the endpoint will be taken offline and considered unreachable. Can be set to a value which is the msec threshold. Setting to no will turn this off. Can also be helpful to keep NAT pinholes open.

Asterisk Trunk Dial Options

Asterisk Dial command options to be used when calling out this trunk. To override the Advanced Settings default, check the box and then provide the required options for this trunk.

Context

(Experts Only) Set the context that calls will originate from. Leaving this as from-internal unless you know what you’re doing.

Continue if Busy

Normally the next trunk is only tried upon a trunk being ‘Congested’ in some form, or unavailable. Checking this box will force a failed call to always continue to the next configured trunk or destination even when the channel reports BUSY or INVALID NUMBER.

Codec

Audio Codecs

You can choose specific audio codecs here

Video Codecs

You can choose specific video codecs here

Item

Definition

Basic

Enable Trunk

Check this to disable this trunk in all routes where it is used.

Trunk Name

Descriptive Name for this trunk

Custom Dial String

Define the custom Dial String.  Include the token $OUTNUM$ wherever the number to dial should go.

examples:

CAPI/XXXXXXXX/$OUTNUM$

H323/$OUTNUM$@XX.XX.XX.XX

OH323/$OUTNUM$@XX.XX.XX.XX:XXXX

vpb/1-1/$OUTNUM$

Advanced

Outbound CallerID

CallerID for calls placed out on this trunk

Format: <#######>. You can also use the format: “hidden” <#######> to hide the CallerID sent out over Digital lines if supported (SIP/IAX).

CID Options

Determines what CIDs will be allowed out this trunk. IMPORTANT: EMERGENCY

CIDs defined on an extension/device will ALWAYS be used if this trunk is part of an EMERGENCY Route regardless of these settings.

Allow Any CID:  all CIDs including foreign CIDS from forwarded external calls will be transmitted.

Block Foreign CIDs: blocks any CID that is the result of a forwarded call from off the system. CIDs defined for extensions/users are transmitted.

Remove CNAM: this will remove CNAM from any CID sent out this trunk

Force Trunk CID: Always use the CID defined for this trunk except if part of any EMERGENCY Route with an EMERGENCY CID defined for the extension/device. Intra-Company Routes will always transmit an extension’s internal number and name.

Maximum Channels

Controls the maximum number of outbound channels (simultaneous calls) that can be used on this trunk. Inbound calls are not counted against the maximum. Leave blank to specify no maximum.

Asterisk Trunk Dial Options

Asterisk Dial command options to be used when calling out this trunk. To override the Advanced Settings default, check the box and then provide the required options for this trunk

Context

(Experts Only) Set the context that calls will originate from. Leaving this as from-internal unless you know what you’re doing.

Continue if Busy

Normally the next trunk is only tried upon a trunk being ‘Congested’ in some form, or unavailable. Checking this box will force a failed call to always continue to the next configured trunk or destination even when the channel reports BUSY or INVALID NUMBER.

Item

Definition

Basic

Description

Provide a meaningful description of what this incoming route is

DID Number

Define the expected DID Number if your trunk passes DID on incoming calls.

Leaving this blank to match calls with any or no DID info.

You can also use a pattern match (eg_2[345]X) to match a range of numbers.

CallerID Number

Define the CallerID Number to be matched on incoming calls.

Leave this field blank to match any or no CID info. In addition to standard dial sequences, you can also put Private, Blocked, Unknown, Restricted, Anonymous and Unavailable in order to catch these special cases if the Telco transmits them.

CID Priority Route

This effects CID ONLY routes where no DID is specified. If checked, calls with this CID will routed to this route, even if there is a route to the DID that was called. Normal behavior is for the DID route to take the calls. If there is a specific DID/CID route for this CID, that route will still take the call when that DID is called.

Inbound Destination

Indicates extension, Ring Group, Voicemail or other destination to which the call is suppoesd to be directed when the outside callers have called specified DID Number

Advanced

Alert Info

ALERT_INFO can be used for distinctive ring with SIP devices.

CID name prefix

You can optionally prefix the CallerID name. ie: If you prefix with “Sales:”, a call from john Doe would display as “Sales: John Doe” on the extension that ring

Music On Hold

Set the MoH class that will be used for calls that come in on this route. For example, choose a type appropriate for routes coming in from a country which may have announcements in their language.

Signal RINGING

Some devices or providers require RINGING to be sent before ANSWER. You’ll notice this happening if you can send calls directly to a phone, but if you send it to an IVR, it won’t connect the call.

Pause Before Answer

An optional delay to wait before processing this route. Setting this value will delay the channel from answering the call. This may be handy if external fax equipment or security systems are installed in parallel and you would like them to be able to seize the line.

Privacy Manager

If no CallerID has been received, Privacy Manager will ask the caller to enter their phone number. If an user/extension has Call Screening enabled, the incoming caller will be prompted to say their name when the call reaches the user/extension.

Source

Source can be added in Caller Name Lookup Sources section.

Language

Allows you to set the language for this DID.

Fax Detect

Attempt to detect faxes on this DID.

Ÿ   No: No attempts are made to auto-determine the call type; all calls sent to destination below. Use this option if this DID is used exclusively for voice OR fax.

Ÿ   Yes: try to auto determine the type of call; route to the fax destination if call is a fax, otherwise send to regular destination. Use this option if you receive both voice and fax calls on this line.

Item

Definition

Basic

Route Name

Name of this route. Should be used to describe what type of calls this route matches (for example, 'local' or 'longdistance').

Route CID

Optional Route CID to be used for this route. If set, this will override all CIDS specified except:

l  extension/device EMERGENCY CIDs if this route is checked as an EMERGENCY Route

l  trunk CID if trunk is set to force it's CID

l  Forwarded call CIDs (CF, Follow Me, Ring Groups, etc)

l  Extension/User CIDs if checked

Route Password

Optional: A route can prompt users for a password before allowing calls to progress.  This is useful for restricting calls to international destinations or 1-900 numbers.

A numerical password, or the path to an Authenticate password file can be used.

Leave this field blank to not prompt for password.

Dial Patterns that will use this Route

A Dial Pattern is a unique set of digits that will select this route and send the call to the designated trunks. If a dialed pattern matches this route, no subsequent routes will be tried. If Time Groups are enabled, subsequent routes will be checked for matches outside of the designated time(s).

Rules:

X  matches any digit from 0-9

Z  matches any digit from 1-9

N  matches any digit from 2-9

[1237-9]  matches any digit in the brackets (example: 1,2,3,7,8,9). wildcard, matches one or more dialed digits

Prepend:   Digits to prepend to a successful match. If the dialed number matches the patterns specified by the subsequent columns, then this will be prepended before sending to the trunks.

Prefix:  Prefix to remove on a successful match. The dialed number is compared to this and the subsequent columns for a match. Upon a match, this prefix is removed from the dialed number before sending it to the trunks.

Match pattern:  The dialed number will be compared against the prefix + this match pattern. Upon a match, the match pattern portion of the dialed number will be sent to the trunks.

CallerID: If CallerID is supplied, the dialed number will only match the prefix + match pattern if the CallerID being transmitted matches this. When extensions make outbound calls, the CallerID will be their extension number and NOT their Outbound CID. The above special matching sequences can be used for CallerID matching similar to other number matches.

Dial patterns wizards

These options provide a quick way to add outbound dialing rules. Follow the prompts for each.

Lookup local prefixes This looks up your local number on ww.localcallingguide.com (NA-only), and sets up so you can dial either 7, 10 or 11 digits (5551234, 6135551234, 16135551234) to access this route.

Upload from CSV Upload patterns from a CSV file replacing existing entries. If there are no headers then the file must have 4 columns of patterns in the same order as in the GUI. You can also supply headers: prepend, prefix, match pattern and callerid in the first row. If there are less than 4 recognized headers then the remaining columns will be blank.

Add Trunks

Trunks used by this outbound route, the available trunks are on the left options bar and the selected on the right.

Advanced

Route Type

Optional: Selecting Emergency will enforce the use of a device

Music On Hold

You can choose which music category to use. For example, choose a type appropriate for a destination country which may have announcements in the appropriate language.

Time Group

If this route should only be available during certain times then Select a Time Group created under Time Groups. The route will be ignored outside of times specified in that Time Group. If left as default of Permanent Route then it will always be available.

Route Position

Where to insert this route or relocate it relative to the other routes.

PIN Set

Optional: Select a PIN set to use. If using this option, leave the Route Password field blank.

Optional Destination on Congestion

If all the trunks fail because of Asterisk ‘CONGESTION’ dial status you can optionally go to a destination such as a unique recorded message or anywhere else. This destination will NOT be engaged if the trunk is reporting busy, invalid numbers or anything else that would imply the trunk was able to make an ‘intelligent’ choice about the number that was dialed. The ‘Normal Congestion’ behavior is to play the ‘ALL Circuits Busy’ recording or other options configured in the route Congestion Messages module when installed.

Item

Definition

Name

Name of this blacklist rule.

Type

Which type the rule applies to, including Inbound/Outbound/Both

Number

Enter the number you want to block, you can input sets of digits that match the Dial Pattern Rules.

Item

Definition

Name

Name of this whitelist rule.

Type

Which type the rule applies to, including Inbound/Outbound/Both

Number

Enter the number you want to add into whitelist, you can input sets of digits that match the Dial Pattern Rules.

Item

Definition

Description

Provide a title for it

CallerID Name 

The callserID name will be changed to it.

CallerID Number 

The callserID number will be changed to it.

Destination 

Destination the call will be sent to after CID has been processed.

Item

Definition

Feature Code Index

There are a total of 10 Feature code objects,0-9, each can control a call flow and be toggled using the call flow toggled feature code plus the index

Name

Description for this Call Flow Toggle Control

Current Mode

This will change the current state for this Call Flow Toggle Control, or set the initial state when creating a new one.

Recording for Normal Mode

Message to be played in normal mode (Green/BLF off)

To add additional recordings use the “System Recordings” MENU to the left

Recording for Override Mode

Message to be played in override mode (Green/BLF off)

To add additional recordings use the “System Recordings” MENU to the left

Optional Password

You can optionally include a password to authenticate before toggling the call flow. If left blank anyone can use the feature code and it will be un-protected

Normal Flow (Green/BLF off)

Destination to use when set to Normal Flow (Green/BLF off) mode

Override Flow (Red/BLF on)

Destination to use when set to Override Flow (Red/BLF off) mode

Item

Definition

Time Condition name

Give this Time Condition a brief name to help you identify it.

Time Group

Select a time group created under Time Groups. Matching times will be sent to matching destination. If no group is selected, call will always go to no-match destination.

Destination if time matches

The destination the call will be sent to when the time matches.

Destination if time does not match 

The destination the call will be sent to when the time doesn’t match.

Item

Definition

Name

It will display as the name of time group

Time

Choose a time interval.

Week Day Start To Finish

Start and end times of one week.

Month Day Start To Finish

Start and end times of one month.

Month Start To Finish

Start and end times of one year.

Item

Definition

Name

Name of the pin sets.

Record In CDR

Select this box if you would like to record the PIN in the call detail records when used.

PIN List

Enter a list of one more PINs. One PIN per line.

Item

Definition

Channel

The FXO Channel number to map to a DID

Description

A useful description this channel

DID

The DID that this channel represents. The incoming call on this channel will be treated as if it came in with this DID and can be managed with Inbound Routing on DIDs

Item

Definition

Delete Used Records

If enabled, when an AutoCLIP record is matched, it will be automatically deleted afterwards.

Record Keep Time

This sets how long each record will be kept in the AutoCLIP List.

Only Keep Missed Call Records

If enabled, the system will only keep records of calls that are not answered by the called party in the AutoCLIP list.

Note: PSTN line will keep records of all calls whether this option is enabled or disabled.

Digits Match

Define how many digits from the last digit of the incoming phone number will be used to match the AutoCLIP record. If the number has fewer digits than the value defined here, all digits will be matched.

Match Outgoing Trunk

If enabled, only the incoming call that came to the PBX through the same trunk which made the call will be match against the AutoCLIP List.

Member Trunks

This defines AutoCLIP Route will apply to which trunk and which trunk's record will be kept in the AutoCLIP list. If no trunk's selected, AutoCLIP will stop working.

Item

Definition

Basic

IVR Name

Name of this IVR

Prompt

The prompt will be played when a call reaches the IVR.

Prompt Repeat Count 

The number of times that the prompt will be played.

Response Timeout (s) 

The number of seconds to wait for a digit input after prompt.

Dial Extensions 

Allow the caller to dial extension directly.

Dial to Check Voicemail

If enabled, the caller will be allowed to dial '*97' to check voicemail.

Advanced

Invalid Retries

Number of time to retry when receiving an invalid/unmatched response from the caller

Invalid Retry Recording

Prompt to be played when an invalid/unmatched response is received, before prompt the caller to try again

Append Announcement on Invalid

After playing Invalid Retry Recording the system will replay mail IVR Announcement

Return on Invalid

Check this box to have this option return to a parent IVR if it was called from a parent IVR. If not, it will go to the chosen destination.

The return path will be to any IVR that was in the call path prior to this IVR which could lead to strange result if there was an IVR called in the call path but not immediately before this.

Invalid Recording

Prompt to be played before sending the caller to an alternate destination due to the caller pressing 0 or receiving the maximum amount of invalid/unmatched responses (as determined by Invalid Retries)

Timeout Retry Recording

Prompt to be played when a timeout occurs, before prompting the caller to try again

Append Announcement on Timeout

After playing the Timeout Retry Recording the system will replay the main IVR Announcement.

Return on Timeout

Check this box to have this option return to a parent IVR if it was called from a parent IVR. If not, it will go to the chosen destination.

The return path will be to any IVR that was in the call path prior to this IVR which could lead to strange result if there was an IVR called in the call path but not immediately before this

Timeout Recording

Prompt to be played before sending the caller to an alternate destination due to the caller pressing 0 or receiving the maximum amount of invalid/unmatched responses (as determined by Invalid Retries)

Return to IVR after VM

If checked, upon exiting voicemail a caller will be returned to this IVR if they got a user voicemail

Item

Definition

Basic

Queue Number

Use this number to dial into the queue, or transfer callers to this number to put them into the queue.

Agents will dial this queue number plus* to log the queue, and this queue number plus** to log out the queue.

For example, if the queue number is 123:

123*=log in

123**=log out

Queue Name

Give the queue a brief name to help you identify it.

Queue Password

You can require agents to enter a password before they can log in to this queue.

This setting is optional.

The password is only used when logging in with the legacy queue no* code. When using the toggle codes, you must use the Restrict Dynamic Agents option in conjunction with the Dynamic Members list to control access.

Generate Device Hints

If checked, individual hints and dialplan will be generated for each SIP and IAX2 device that could be part of this queue. These are used in conjunction with programmable BLF status as to the current state, the format of this hints is

*45ddd*qqq

Where *45 is the currently define toggle feature code, ddd is the device number (typically the same as the extension number) and qqq is this queue’s number

Call Confirm

If checked, any queue member that is actually an outside telephone number, or any extension Follow-Me or call forwarding that are pursued and leave the PBX will be forced into Call Confirmation mode where the member must acknowledge the call before it is answered and delivered.

Call Confirm Announce

Announcement played to the Queue Member announcing the Queue call and requesting confirmation prior to answering. If set to default, the standard call confirmation default message will be played unless the number is reached through a Follow-Me and this is an alternate message provided in the Follow-Me. This message will override any other message specified.

To add additional recordings please use the “System Recordings” MENU.

CID Name Prefix

You can optionally prefix the CallerID name of callers to the queue. ie: If you prefix with “Sales:”, a call from John Doe would display as “Sales: John Doe” on the extensions that ring.

Wait Time Prefix

When set to Yes, the CID Name will be prefix with the total wait time in the queue so the answering agent is aware how long they have waited. It will be rounded to the nearest minute, in the form of Mnn: where nn is the number of minutes.

If the call is subsequently transferred, the wait time will reflect the time since it first entered the queue or reset if the call is transferred to another queue with this feature set.

Alert Info

ALERT_INFO can be used for distinctive ring with SIP device.

Static Agents

Static agents are extensions that are assumed to always be on the queue.  Static agents do not need to 'log in' to the queue, and cannot 'log out' of the queue.

List extensions to ring, one per line.

You can include an extension on a remote system, or an external number (Outbound

Routing must contain a valid route for external numbers). You can put a "," after the agent followed by a penalty value, see Asterisk documentation concerning penalties.

An advanced mode has been added which allows you to prefix an agent number with S, X, Z, D or A. This will force the agent number to be dialed as an Asterisk device of type SIP, IAX2, ZAP, DAHDi or Agent respectively. This mode is for advanced users and can cause known issues in PBX as you are by-passing the normal dialplan. If your 'Agent Restrictions' are not set to 'Extension Only' you will have problems with subsequent transfers to voicemail and other issues may also exist.

(Channel Agent is deprecated starting with Asterisk 1.4 and gone in 1.6+.)

Dynamic Members

Dynamic Members are extensions or callback numbers that can log in and out of the queue. When a member logs in to a queue, their penalty in the queue will be as specified here. Extensions included here will NOT automatically be logged in to the queue.

Restrict Dynamic Agents

Restrict dynamic queue member logins to only those listed in the Dynamic Members list above. When set to Yes, members not listed will be DENIED ACCESS to the queue.

Agent Restrictions

When set to ‘Call as Dialed’ the queue will call an extension just as if the queue were another user. Any Follow-Me or Call Forward states active on the extension will result in the queue call following these call paths. This behavior has been the standard queue behavior on past PBX versions.

When set to ‘No Follow-Me or Call Forward’, all agents that are extensions on the system will be limited to ring their extensions only. Follow-Me and Call Forward settings will be ignored. Any other agent will be called as dialed. This behavior is similar to how extensions are dialed in ringgroups

When set to ‘Extensions Only’ the queue will dial Extensions as described for ‘No Follow –Me or Call Forward’. Any other number entered for an agent that is NOT a valid extension will be ignored. No error checking is provided when entering a static agent or when logging on as a dynamic agent, the call will simply be blocked when the queue tries to call it. For dynamic agents, see the ‘Agent Regex filter’ to provide some validation.

General Queue Options

Ring Strategy

Ringall: ring all available agents until one answers (default)

Leastrecent: ring agent which was least recently called by this queue

Fewestcalls: ring the agent with fewest completed calls from this queue

Random: ring random agent

Rrmemory: round robin with memory, remember where we left off last ring pass

Rrordered: same as rrmemory, except the queue member where order from config file is preserved

Linear: rings agents in the order specified, for dynamic agents in the order they logged in

Wrandom: random using the member’s penalty as a weighting factor, see asterisk documentation for specifics.

Autofill

Starting with Asterisk 1.4, if this is checked, and multiple agents are available, Asterisk will send one call to each waiting agent(depending on the ring strategy). Otherwise, it will hold all calls while it tries to find an agent for the top call in the queue making other calls wait. This was the behavior in Asterisk 1.2 and has no effect in 1.2. See Asterisk documentation for more details of this feature.

Skip Busy Agents

When set to 'Yes' agents who are on an occupied phone will be skipped as if the line were returning busy. This means that Call Waiting or multi-line phones will not be presented with the call and in the various hunt style ring strategies, the next agent will be attempted.

When set to 'Yes + (ringinuse=no)' the queue configuration flag 'ringinuse=no' is set for this queue in addition to the phone's device status being monitored. This results in the queue tracking remote agents (agents who are a remote PSTN phone, called through Follow-Me, and other means) as well as PBX connected agents, so the queue will not attempt to send another call if they are already on a call from any queue.

When set to 'Queue calls only (ringinuse=no)' the queue configuration flag 'ringinuse=no' is set for this queue also but the device status of locally connected agents is not monitored. The behavior is to limit an agent belonging to one or more queues to a single queue call. If they are occupied from other calls, such as outbound calls they initiated, the queue will consider them available and ring them since the device state is not monitored with this option.

WARNING: When using the settings that set the 'ringinuse=no' flag, there is a NEGATIVE side effect. An agent who transfers a queue call will remain unavailable by any queue until that call is terminated as the call still appears as 'inuse' to the queue UNLESS 'Agent Restrictions' is set to 'Extensions Only'.

Queue Weight

Gives queue a ‘weight’ option, to ensure calls waiting in a higher priority queue will deliver its calls first if there are agents common to both queues.

Music on Hold Class

Music (MoH) played to the caller while they wait in line for an available agent. Choose “inherit” if you want the MoH class to be what is currently selected, such as by the inbound route. MoH Only will play music until the agent answers. Agent Ringing will play MoH until an agent’s phone is presented with the call and is  ringing. If they don’t answer MoH will return. Ring only makes callers hear a ringing tone instead of MoH ignoring any MoH class selected as well as any configured periodic announcements. This music is defined in the “Music on Hold” Menu.

Join Announcement

Announcement played to callers prior to joining the queue. This can be skipped if there are agents ready to answer a call (meaning they still may be wrapping up from a previous call) or when they are free to answer the call right now. To add additional recordings please use the “System Recordings” MENU.

Caller Volume Adjustment

Adjust the recording volume of the caller.

Agent Volume Adjustment

Adjust the recording volume of the queue member (Agent).

Mark calls answered elsewhere

Enabling this option, all calls are marked as ‘answered elsewhere’ when cancelled. The effect is that missed queue calls are *not* shown on the phone(if the phone support it)

Timing & Agent Options

Max Wait Time

The maximum number of seconds a caller can wait in a queue before being pulled out.(0 for unlimited).

Max Wait Time Mode

Asterisk timeoutpriority. In ‘Strict’ mode, when the ‘Max Wait Time’ of a caller is hit, they will be pulled out of the queue immediately. In ‘Loose’ mode, if a queue stops ringing with this call, then we will wait until the queue stops ringing this queue number or otherwise the call is rejected by the queue member before taking the caller out of the queue. This means that the ‘Max Wait Time’ could be as long as ‘Max Wait Time’+’Agent Timeout’ combined.

Agent Timeout

The number of seconds an agent’s phone can ring before we consider it a timeout. Unlimited or other timeout values may still be limited by system ringtime or individual extension defaults.

Agent Timeout Restart

If timeout restart is set to yes, then the time out for an agent to answer is reset if a BUSY or CONGESTION is received. This can be useful if agents are able to cancel a call with reject or similar

Retry

The number of seconds we wait before trying all the phones again. Choosing “No Retry” will exit the queue and go to the fail-over destination as soon as the first attempted agent time-out, additional agents will not be attempted.

Wrap-Up-Time

After a successful call, how many seconds to wait before sending a potentially free agent another call (default is 0, or no delay) If using Asterisk 1.6+, you can also set the ‘Honor Wrapup Time Across Queues setting (Asterisk: shared_lastcall) on the Advanced Settings page so that this is honored across queues for members logged on to multiple queues.

Member Delay

If you wish to have a delay before the member is connected to the caller (or before the member hears any announcement messages), set this to the number of seconds to delay.

Agent Announcement

Announcement played to the Agent prior to bridging in the caller.

Example : ”the Following call is from the Sales Queue” or “This call is from the Technical Support Queue”.

To add additional recordings please use the “System Recordings” MENU. Compound recordings composed of 2 or more sound files are not displayed as options since this feature can not accept such recordings.

Report Hold Time

If you wish to report the caller’s hold time to the member before they are connected to the caller, set this to yes.

Auto Pause

Auto Pause an agent in this queue (or all queues they are a member of ) if they don’t answer a call. Specific behavior can be modified by the Auto Pause Delay as well Auto Pause Busy/Unavailable settings if supported on this version of Asterisk.

Auto Pause on Busy

When set to Yes agents devices that report busy upon a call attempt will be considered as a missed call and auto paused immediately or after the auto pause delay if configured

Auto Pause on Unavailable

When set to Yes agents devices that report congestion upon a call attempt will be considered as a missed call and paused immediately or after that auto pause delay if configured

Auto Pause Delay

This setting will delay the auto pause of an agent by auto pause delay seconds from when it last took a call. For example, if this were set to 120 seconds, and a new call is presented to the agent 90 seconds after they last took a call, will not be auto paused if they don’t answer the call. If presented with a call 120 seconds or later after answering the last calls, this will have no effect.

Capacity Options

Max Callers

Maximum number of people waiting in the queue (0 for unlimited)

Join Empty

Determines if new callers will be admitted to the Queue, if not, the failover destination will be immediately pursued. The options include:

l  Yes Always allows the caller to join the Queue.

l  Strict Same as Yes but stricter. Simply speaking, if no agent could answer the phone then don’t admit them. If agents are infused or ringing someone else, caller will still be admitted.

l  Ultra Strict Same as Strict plus a queue member must be able to answer the phone ‘now’ to let them in. simply speaking, any ‘available’ agents that could answer but are currently on the phone or ringing on behalf of another caller will be considered unavailable.  

l  No Callers will not be admitted if all agents are paused, show an invalid status for their device, or have penalty values less than QUEUE_MAX_PENALTY (not currently set in dialplan).

l  Loose Same as No except Callers will be admitted if there are paused agents who could become available.

Leave Empty

Determines if callers should be exited prematurely from the queue in situations where it appears no one is currently available to take the call. The options include:

l  Yes Callers will exit if all agents are paused, show an invalid state for their device or have penalty values less than QUEUE_MAX_PENALTY (not currently set in dialplan)

l  Strict Same as Yes but stricter. Simply speaking, if no agent could answer the phone then have them leave the queue. If agents are inuse or ringing someone else, caller will still be held.

l  Ultra Strict Same as Strict plus a queue member must be able to answer the phone ‘now’ to let them remain. simply speaking, any ‘available’ agents that could answer but are currently on the phone or ringing on behalf of another caller will be considered unavailable.  

l  Loose Same as No except Callers will remain in the queue, if there are paused agents who could become available.

l  No never have a caller leave the Queue until the Max Wait Time has expired.

Penalty Members Limit

Asterisk: penalty members limit. A limit can be set to disregard penalty settings, allowing all members to be tried, when the queue has too fewer members. No penalty will be weight in if there are only X or fewer queue members.

Frequency

How often to announce queue position and estimated holdtime (0 to Dis able Announcements).

Announce Position

Announce position of caller in the queue

Announce Hold Time

Should we include estimated hold time in position announcements? Either yes, no, or only once; hold time will not be announced if <1 minute.

IVR Break Out Menu

You can optionally present an existing IVR as a ‘break out’ menu.

This IVR must only contain single-digit ‘dialed options’ . The recording set for the IVR will be played at intervals specified in ‘Repeat Frequency’, below.

Repeat Frequency

How often to announce a voice menu to the caller (0 disable Announcements)

Event When Called

When this option is set to YES, the following manager events will be generated: AgentCalled, AgentDump, AgentConnect and AgentComplete.

Member Status Event

When set to YES, the following manager event will be generated: QueueMemberStatus.

Service Level

Used for service level statistics (calls answered within service level time frame)

Agent Regex Filter

Provides an optional regex expression that will be applied against the agent callback number. If the callback number does not pass the regex filter then it will be treated as invalid. This can be used to restrict agents to extensions within a range, not allow callbacks to include keys like *, or any other use that may be appropriate. An example input might be:

^([2-4][0-9]{3})$

This would restrict agents to extensions 2000-4999. Or

^([0-9]+)$ would allow any number of any length, but restrict the * key.

WARNING: make sure you understand what you are doing or otherwise leave this blank!

Run

Select how often to reset queue stats. The following schedule will be followed for all but custom:

Hourly Run once an hour, beginning of hour

Daily Run once a day, at midnight

Weekly Run once a week, midnight on Sun

Monthly Run once a month, midnight, first of month

Annually Run once a year, midnight, Jan.1

Reboot Run at startup of the server OP of the cron deamon (i.e. after every service cron restart)

If Randomize is selected, a similar frequency will be followed, only the exact times will be randomized (avoiding peak business hours, when possible). Please note: randomized schedules will be rescheduled (randomly) every time ANY backup is saved.

Item

Definition

Speed Dial

This option must be checked

Name

Name of the speed dial

Number

Destination external number

Speed dial code

A number to associate this code to the external number to dial

Item

Definition

DISA name

Give this DISA a brief name to help you identify it.

PIN

The user will be prompted for this number. If you wish to have multiple PIN’s, separate them with commas.

Response Timeout

The maximum amount of time it will before hanging up if the user has dialed an incomplete or invalid number. Default of 10 seconds.

Digit Timeout

The maximum amount of time permitted between digits when the user is typing in an extension. Default of 5.

Require Confirmation

Require Confirmation before prompting for password. Used when your PSTN connection appears to answer the call immediately.

Caller ID

(Optional) When using this DISA, the users CallerID will be set to this. Format is “User Name” <5551234>

Context

(Experts Only) Set the context that calls will originate from. Leaving this as from-internal unless you know what you’re doing.

Allow Hangup

Allow the current call to be disconnected and dial tone presented for a new call by pressing the Handup feature code: ** while in a call.

Caller ID Override

Determine if we keep the Caller ID being presented or if we override it. Default is Enable.

Item

Definition

Basic

Conference Number

Use this number to dial into the conference.

Conference Name

Give this conference a brief name to help you identify it.

User PIN

You can require callers to enter a password before they can enter this conference.

This setting is optional.

If either PIN is entered, the user will be prompted to enter a PIN.

Admin PIN

Enter a PIN number for the admin user.

This setting is optional unless the ‘leader wait’ option is in use, then this PIN will identify the leader.

Advanced

Join Message

Message to be played to the caller before joining the conference.

To add additional recordings use the “System Recordings” MENU to the left

Leader Wait

Wait until the conference leader (admin user) arrives before starting the conference

Talker Optimization

Turn on talker optimization. With talker optimization, Asterisk treats talkers who are not speaking as being muted, meaning that no encoding is done on transmission and that received audio that is not registered as talking is omitted, causing no buildup in background noise.

Talker Detection

Sets talker detection. Asterisk will sends events on the Manager Interface identifying the channel that is talking. The talker will also be identified on the output of the meetme list CLT command.

Quiet Mode

Quiet mode (do not play enter/leave sounds)

User Count

Announce user(s) count on joining conference

User join/leave

Announce user join/leave

Music on Hold

Enable Music on Hold when the conference has single caller

Music on Hold Class

Music (or Commercial) played to the caller while they wait line for the conference to start. Choose “inherit” if you want the MoH class to be what is currently selected, such as by the inbound route.

This music is defined in the “Music on Hold” to the left.

Allow Menu

Present Menu (user or admin) when ‘*’ is received (‘send’ to menu).

Record Conference

Record the conference call

Maximum Participants

Maximum Number of users allowed to join this conference.

Mute on Join

Mute everyone when they initially join the conference. Please note that if you do not have ‘Leader Wait’ set to yes you must have ‘Allow Menu’ set to Yes to unmute yourself.

Item

Definition

Callback Description

Enter a description for this callback

Callback Number

Optional: Enter the number to dial for the callback. Leave this blank to just dial the incoming CallerID Number.

Delay Before Callback

Optional: Enter the number of seconds the system should wait before calling back.

Destination

Destination of a callback.

Item

Definition

Callback Description

Enter a description for this callback

Callback Number

Optional: Enter the number to dial for the callback. Leave this blank to just dial the incoming CallerID Number.

Delay Before Callback

Optional: Enter the number of seconds the system should wait before calling back.

Item

Definition

Basic

Parking Lot Extension

This is the extension where you will transfer a call to park it

Parking Lot Name

Name of the parking Lot.

Parking Lot Starting Position

The starting postion of the parking lot.

Number of Slots

The total number of parking lot spaces to configure.

Parking Timeout (seconds)

The timeout period in seconds that a parked call will attempt to ring back the original parker if not answered.

Parked Music Class

This is the music class that will be played to a parked call while in the parking lot UNLESS the call flow prior to parking the call explicitly set a different music class

BLF Capabilities

Enable this to have Asterisk “hints” generated to use with BLF buttons.

Find Slot

If you want the parking lot to seek the next sequential parking slot relative to the the last parked call instead of seeking the first available slot.

Destination

Destination of Parking Lot.

Advanced

Pickup Courtesy Tone

Whom to play the courtesy tone to when a parked call is retrieved.

CallerID Prepend

String to prepend to the current Caller ID associated with the parked call prior to sending back to the Originator or the Alternate Destination.

Transfer Capability

parkedcalltransfers. Enables or disables DTMF based transfers when picking up a parked call.

Parking Alert-Info

Alert-Info to add to the call prior to sending back to the Originator or to the Alternate Destination.

Re-Parking Capability

parkedcallreparking. Enables or disables DTMF based parking when picking up a parked call

Auto CallerID Prepend

These options will be appended after CallerID Prepend if set.

Announcement

Optional message to be played to the call prior to send back to the originator.

Come Back to Origin

Where to send a parked call that has timed out. If set to yes then the parked call will be sent back to the originating device that sent the call to this parking lot.

*2nn:

Attended Transfer call into Park lot nnn (It will announce the slot back to you)

nn:

Park Yourself into Parking lot nnn (Announcing your parked slot to you)

Item

Definition

VMBlast Number

The number users will dial to voicemail boxes in this VMBlast group

Group Description

Provide a descriptive title for this VMBlast Group.

Audio Label

Paly this message to the caller so they can confirm they have dialed the proper voice mail group number, or have the system simply read the group number.

Optional Password

You can optionally include a password to authenticate before providing access to this group voicemail list.

Voicemail Box List

Select voice mail boxes to add to this group. Use Ctrl key to select multiple.

Default VMBlast Group

Each PBX system cam have a single Default VOICEMAIL Blast Group. If specified, extensions can be automatically added (or removed) from this default group in the Extensions (or Users) tab.

Making this group the default will uncheck the option from the current default group if specified.

Options

Definition

Asterisk Manager

Asterisk Manager Password

Password for accessing the Asterisk Manager Interface (AMI), this will be automatically updated in manager.conf.

Asterisk Manager User

Username for accessing the Asterisk Manager Interface (AMI), this will be automatically updated in manager.conf.

System Setup

Aggressively Check for Duplicate Extensions

Aggressively Check for Duplicate Extensions

User & Devices Mode

Sets the extension behavior in UC device.  If set to extensions, Devices and Users are administered together as a unified Extension, and appear on a single page. If set to deviceanduser, Devices and Users will be administered separately.

Call Recording Format 

Format to save recoreded calls for most call recording unless specified differently in specific applications.

Convert Music Files to WAV 

When set to false, the MP3 files can be loaded and WAV files converted to MP3 in the MoH module. The default behavior of true assumes you have mpg123 loaded as well as sox and will convert MP3 files to WAV. This is highly recommended as MP3 files heavily tax the system and can cause instability on a busy phone system

Voice Prompts

Default language

The default language for Voice Prompts.

Options

Definition

Tone duration

How long generated tones (DTMF and MF) will be played on the channel. (in milliseconds)

Codec

Set the global encoding: ulaw, alaw.

Impedance

Configuration for impedance.

Echo Cancellation 

Enable/Disable

EC Taps

128/256/512/1024

Flash/Wink

Turn on/off Flash/wink.

Min flash time

Min flash time. (in milliseconds).

Max flash time

Max flash time. (in milliseconds).

Ending Dial Key

Turn on/off Ending Dial Key (#).

Hang up on polarity switch

Turn on/off Hangup on polarity switch

Option

Description

The pattern of sending CID

Some countries (UK) have ring tones with 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 (1).

Waiting time before

sending CID

How long we will be waiting before sending the CID on the channel. (in milliseconds).

Sending polarity

reversal (DTMF Only)

Send polarity reversal before sending the CID on the channel.

Start code (DTMF Only)

Start code.

Stop code (DTMF Only)

Stop code.

Options

Definition

Use CallerID

Turn on/off CallerID detect function

Hide CallerID

Turn on/off CallerID detect function

Callerid

Caller ID can be set to "asreceived" or a specific number if you want to override it. Note that "asreceived" only applies to trunk interfaces.

CID Signalling

CID Start 

polarity=polarity reversal signals the start

Options

Definition

Silence detect

Turn on/off silence detect function

Silence threshold

What we consider silence: the lower, the more sensitive, eg:250 is 250ms. Range: 100 to 500(100 to 500ms), default: 250

Silence length

How many silence threshold of silence before hanging up(eg: 16 is 250ms*16=4s). Range: 2 to 1020 (200ms to 512s), default: 80(20s)

Silence framesize

Rx threshold

Range: -20 dBm0 to -40 dBm0, default: 20(-20 dBm0), all values are understood to be negative.

Tx threshold

Range: -20 dBm0 to -40 dBm0, default: 20(-20 dBm0), all values are understood to be negative.

Options

Definition

User Agent 

Value used in User-Agent header for SIP requests and Server header for SIP responses.

Realm

When generates a challenge, the digest realm will be set to this value if there is no better option (such as auth/realm) to be used.

Allow Guests 

When set Asterisk will allow Guest SIP calls and send them to the Default SIP context. Turning this off will keep anonymous SIP calls from entering the system. Doing such will also stop 'Allow Anonymous Inbound SIP Calls' from functioning. Allowing guest calls but rejecting the Anonymous SIP calls below will enable you to see the call attempts and debug incoming calls that may be mis-configured and appearing as guests.

Domain The Transport Comes From

Typically used with SIP calling. Example user@domain, where domain is the value that would be entered here

Local networks 

Local network settings in the form of ip/cidr or ip/netmask. For networks with more than 1 LAN subnets, use the Add Local Network Field button for more fields. Blank fields will be ignored.

External Media Address

External IP address to use in RTP handling.

External Signaling Address

External address for SIP signaling.

External Signaling Port 

External port for SIP signaling.

Options

Definition

UDP

Enable

USE 0.0.0.0 - All

Bind Host

You can customize the UDP bind host, the default is 0.0.0.0

Port To Listen On

The port that this transport should listen on

TCP

Bind Host

You can customize the TCP bind host, the default is 0.0.0.0

Port To Listen On

The port that this transport should listen on

TLS

Bind Host

You can customize the TLS bind host, the default is 0.0.0.0

Port To Listen On 

The port that this transport should listen on

Certificate Manager

Select a certificate to use for the TLS transport. These are configured in the module Certificate Manager1

SSL Method 

Method of SSL transport (TLS ONLY). The default is currently TLSv1, but may change with future releases.1

Verify Client 

Require verification of server certificate (TLS ONLY).

Verify Server

Require verification of server certificate (TLS ONLY).

WS

Bind Host

You can customize the WS bind host, the default is 0.0.0.0

WSS

Bind Host 

You can customize the WSS bind host, the default is 0.0.0.0

Options

Definition

Codec Priority

Asterisk: codecpriority. Controls the codec negotiation of an inbound IAX call. This option is inherited to all user entities. It can also be defined in each user entity separately which will override the setting here. The valid values are:host - Consider the host's preferred order ahead of the caller's.caller - consider callers host's. disabled disable consideration codec preference altogether. (this is original behavior before preferences were added)reqonly same as disabled, only do not capabilities if requested format available call will be accepted available.

Bandwidth

Asterisk: bandwidth. Specify bandwidth of low, medium, or high to control which codecs are used in general.

Video Support

Check to enable and then choose allowed codecs. If you clear each codec and then add them one at a time, submitting with each addition, they will be added in order which will effect the codec priority.

Audio Codecs

Check the desired codecs, all others will be disabled unless explicitly enabled in a device or trunks configuration. Drag to re-order.

Options

Definition

Language

Default Language for a channel, Asterisk: language

Bind Address

Asterisk: bindaddr. The IP address to bind to and listen for calls on the Bind Port. If set to 0.0.0.0 Asterisk will listen on all addresses. To bind to multiple IP addresses or ports, use the Other iax settings' fields where you can put settings such as:bindaddr='192.168.10.100:4555.' it is recommended to leave this blank.

Bind Port

Asterisk: bindport. Local incoming UDP Port that Asterisk will bind to and listen for IAX messages. The IAX standard is 4569 and in most cases this is what you want. It is recommended to leave this blank.

Delay Auth Rejects

Asterisk: delayreject. For increased security against brute force password attacks enable this which will delay the sending of authentication reject for REGREQ or AUTHREP if there is a password.

Other IAX Settings

You may set any other IAX settings not present here that are allowed to be configured in the General section of iax.conf. There will be no error checking against these settings so check them carefully. They should be entered as: [setting] = [value]in the boxes below. Click the Add Field box to add additional fields. Blank boxes will be deleted when submitted.

Options

Definition

Strict RTP

Enable strict RTP protection. This will drop RTP packets that do not come from the source of the RTP stream. This option is disabled by default.

RTP Checksums

Whether to enable or disable UDP checksums on RTP traffic

ICE Support

Whether to enable ICE support. Defaults to no. ICE (Interactive Connectivity Establishment) is a protocol for network address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model. This option is commonly enabled in WebRTC setups

RTP Start

Start of range of port numbers to be used for RTP. Defaults is 10000.

RTP End

End of range of port numbers to be used for RTP. Defaults is 20000.

Reinvite Behavior

yes: standard reinvites; (yes = update + nonat)

no: never;

nonat: An additional option is to allow media path redirection (reinvite) but only when the peer where the media is being sent is known to not be behind a NAT (as the RTP core can determine it based on the apparent IP address the media arrives from;

update: use UPDATE for media path redirection, instead of INVITE.

RTP Time Out

The call is terminated when there is no RTP or RTCP activity on the audio channel for a period of time (that is, the set timeout period). This is to be able to hang up the call in case of network interruption (not on hold)

RTP Hold Time Out

If there is no RTP or RTCP activity on the audio channel for a period of time (that is, the set hold timeout period), the call will be terminated (in hold state). This value must be greater than the timeout period.

RTP Keep Alive

Send Keepalive in RTP stream to keep NAT open (default is off)

STUN Server

Configure the STUN server address. STUN is a Client/Server protocol and also a Request/Response protocol. It is used here to check the connectivity between two terminals, like a way to maintain NAT binding entries Keep-alive agreement.

TURN Server

Configure the TURN server address, STUN can handle most of the NAT problems. TURN is an enhanced version of the STUN protocol, dedicated to dealing with symmetric NAT problems.

TURN Server Name

Configure the TURN Server name

TURN Server Password

Configure the TURN Server password

Item

Definition

Description

Give this Misc Destination a brief name to help you identify it.

Dial

Enter the number this destination will simulate dialing, exactly as you would dial it from an internal phone. When you route a call to this destination, it will be as if the caller dialed this number from an internal phone.

Item

Definition

Manager name

Name of the manager without space.

Manager secret

Password for the manager.

Deny

If you want to deny many hosts or networks, use & char as separator.

Example: 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0

Permit

If you want to permit many hosts or networks, use & char as separator. Look at deny example.

Item

Definition

AMI Port

Sets the port number to listen on for AMI connections. Port numbers can not be less than 1024. The default is 7777.

TLS Enable

Enables listening for AMI connections using TLS. The default is no.

TLS Port

Sets the port to listen on for TLS connections to the AMI. The default is 5039.

TLS Bind Address 

Sets the address to listen on for TLS-based AMI connections. The default is to listen on all addresses (0.0.0.0).

Timestamp Events  

Add a Unix epoch timestamp to events (but not action responses.

Write Timeout 

Write Timeout 

Item

Description

Status

This displays the status of the endpoint as one or more icons. The available flags are as follows:

Scroll icon: the endpoint has not been scanned, but rather defined in an upload.

Disk icon: the endpoint configuration has been updated in the database but not yet applied to its configuration files.

Person icon: the endpoint has at least one endpoint assigned.

MAC Address

This is the main identifier for the endpoint. Configurations in the database and uploaded files are considered to refer to the same endpoint if they reference the same MAC address.

Current IP

If the endpoint was detected through a scan, this field will show the IP at which the endpoint was found. This field is a link to the HTTP configuration interface (if supported) of the phone.

Manufacturer

This displays the detected manufacturer of the endpoint.

Model

This displays the detected model of the endpoint. Since automatic model detection is not (yet) implemented for some manufacturers, this field allows the user to correct the model via a drop-down list. Accurate model detection is required for many other features (such as account assignment) to work.

Options

This link displays a modal dialog on which common options for the endpoint can be manually configured.