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

Menu

Administrator

Operator

Extension

System 

System Info

Yes

Yes

No

PBX Configuration

Yes

No

No

Network

Yes

No

No

User Management

Yes

No

No

Shutdown

Yes

No

No

Operator Panel

Flash Operator Panel

Yes

Yes

No

Voicemails

Asterisk Recording Interface

Yes

Yes

No

Fax

Virtual Fax List

Yes

Yes

No

New Virtual Fax

Yes

No

No

Reports

CDR Report

Yes

Yes

No

Channels Usage

Yes

Yes

No

Billing

Rates

Yes

No

No

Billing Report

Yes

No

No

Destination Distribution

Yes

No

No

Trunk Configuration

Yes

No

No

Extras 

SugarCRM

Yes

Yes

Yes

Calling Cards

Yes

Yes

Yes

Downloads

Softphones

Yes

Yes

Yes

Fax Utilities

Yes

Yes

Yes

Item

Description

Basic

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.

Secret

Password (secret) configured for the device. Should be alphanumeric with at least 2 letters and numbers to keep secure.

Dtmfmode

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

Canreinvite

Re-Invite policy for this device, see Asterisk documentation for details.

Context

Asterisk context this device will send calls to. Only change this is you know what you are doing.

Host

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

Trustrpid

Whether Asterisk should trust the RPID settings from this device. Usually should be yes for CONNECTEDLINE( ) functionality to work if supported by the endpoint.

Sendrpid

Whether Asterisk should send RPID (or PAI) info to the device. Usually should be enabled to the settings used by your device for CONNECTEDLINE( ) functionality to work if supported by the endpoint.

The PRACK request plays the same role as ACK, but for provisional responses.

Type

Asterisk connection type, usually friend for endpoint.

NAT

NAT seting, see Asterisk documentation for details. Yes usually works for both internal and external devices. Set to No if the device will always be internal.

Port

Endpoint port number to use, usually 5060. Some 2 ports devices such as ATA may used 5061 for the second port.

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.

Qualifyfreq

Frequency in seconds to send qualify messages to the endpoint.

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.

Avpf

Whether to Enable AVPF. Defaults to no. the WebRTC standard has selected AVPF as the audio video profile to use for media streams. This is not the default profile in use by Asterisk. As a result the following must enabled to use WebRTC.

Icesupport

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.

Dtlsenable

Whether to enable DTLS for this peer. Defaults to no.

Dtlsverify

Whether to verify that the provided peer certificate is valid. Defaults to no.

Dtlssetup

Behavior on DTLS incoming and outgoing connections, defaults to actpass.

Dtlscertfile

Path to certificate file to present.

Dtlscafile

Path to cafile file to present

Dtlsprivatekey

Path to private key for certificate file.

Encryption

Whether to offer SPTR encrypted media (and only SRTP encrypted media) on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if the peer does not support SRTP. Defaults to no.

Callgroup

Callgroup(s) that this device is part of, can be or more callgroups, e.g. '1,3-5' would be in groups 1,3,4,5.

Pickupgroup

Pickupgroup(s) that this device can pickup calls from, can be one or more groups, e.g. '1.3-5' would be in groups 1,3,4,5. Device does not have to be in a group to be able to pickup calls from that group.

Allow

Dial

How to dial device, this should not be changed unless you know what you are doing.

Accountcode

Accountcode for this device.

Mailbox

Mailbox for this device. This should not be changed unless you know what you are doing.

Vmexten

Asterisk dialplan extension to reach voicemail for this device. Some devices use this to auto-program the voicemail button on the endpoint. If left blank, the default vmexten setting is automatically configured by the voicemail module. Only changed this on devices that may have special needs.

Deny

IP Address range to deny access to, in the form of network/netmask.

Permit

IP Address range to allow access to, in the form of network/netmask. This can be a very useful security option when dealing with remote extensions that are at a known location (such as a branch office) or with a known ISP range for some home office situations.

Email Address

The email address that completed dictations are sent to.

Language Code

This will cause all messages and voice prompts to use the selected language if installed.

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. A common example is a team of support people who would like their internal CallerID to display the general support number(a ringgroup or queue). There will be no effect on external calls.

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 a 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.

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 affects 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'.

DID Description

A description for this DID, such as "Fax".

Add Inbound DID

A direct DID that is associated with this extension. The DID should be in the same format as provider (e.g. full number, 4digits for 10x4, etc).

Add Inbound CID

Add a CID for more specific DID+CID routing. A DID must be specified in the above Add DID box. 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.

Recording

Inbound External Calls

Recording of inbound calls from external sources.

Outbound External Calls

Recording of outbound calls from external sources.

Inbound Internal Calls

Recording of calls received from other extensions on the system.

Outbound Internal Calls

Recording of calls made to other extensions on the system.

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 can not 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.

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.

Email Address

The email address that Voicemails are sent to.

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.

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

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 FreePBX 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.

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:

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

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 FreePBX DND, it won't be rung. If the primary is FreePBX 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

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

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.

Outbound Proxy Server

Example: proxy.provider.domain:port

Host

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

Username

Secret

Password (secret) configured for the device. Should be alphanumeric with at least 2 letters and numbers to keep secure.

Port

Endpoint port number to use, usually 5060. Some 2 ports devices such as ATA may used 5061 for the second port.

Register String

Most VoIP providers require your system to REGISTER with theirs. Enter the registration line here.
example:
username:password@switch.voipprovider.com.
Many providers will require you to provide a DID number, ex: username:password@switch.voipprovider.com/didnumber in order for any DID matching to work.

Codec

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

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 the auto-generated context: as the inbound trunk's context. (see extensions_additional.conf) Leave blank to specify no maximum.

Type

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.

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.

Qualifyfreq

Frequency in seconds to send qualify messages to the endpoint.

Nat

NAT seting, see Asterisk documentation for details. Yes usually works for both internal and external devices. Set to No if the device will always be internal.

Insecure

Specifies how to handle connections with peers. Default no (authenticate all connections).

Dtmfmode

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

Trustrpid

Whether Asterisk should trust the RPID settings from this device. Usually should be yes for CONNECTEDLINE( ) functionality to work if supported by the endpoint.

Sendrpid

Whether Asterisk should send RPID (or PAI) info to the device. Usually should be enabled to the settings used by your device for CONNECTEDLINE( ) functionality to work if supported by the endpoint.

Prack

The PRACK request plays the same role as ACK, but for provisional responses.

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 messages 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

Avpf

Whether to Enable AVPF. Defaults to no. the WebRTC standard has selected AVPF as the audio video profile to use for media streams. This is not the default profile in use by Asterisk. As a result the

Icesupport

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

Dtlsenable

Whether to Enable DTLS for this peer. Defaults to no.

Dtlsverify

Whether to verify that the provided peer cerificate is valid. Defaults to no.

Dtlssetup

Behavior on DTLS incoming and outgoing connections. Defaults to actpass.

Dtlscertfile

Path to certificate file to present

Dtlscafile

Path to cafile file to present

Dtlsprivatekey

Path to private key for certificate file.

Encryption

Whether to offer SRTP encrypted media (and only SRTP encrypted media) on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if the peer does not support SRTP. Defaults to no.

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

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

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

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.

Username configured for this trunk.

Secret

Password (secret) configured for the device. Should be alphanumeric with at least 2 letters and numbers to keep secure.

Port

Endpoint port number to use, usually 5060. Some 2 ports devices such as ATA may used 5061 for the second port.

Register String

Most VoIP providers require your system to REGISTER with theirs. Enter the registration line here.
example:
username:password@switch.voipprovider.com.
Many providers will require you to provide a DID number, ex: username:password@switch.voipprovider.com/didnumber in order for any DID matching to work.

Codec

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

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.

Type

Asterisk connection type, usually friend for endpoint.

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.

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.

Qualifyfreq

Frequency in seconds to send qualify messages to the endpoint.

Nat

NAT seting, see Asterisk documentation for details. Yes usually works for both internal and external devices. Set to No if the device will always be internal.

Insecure

Specifies how to handle connections with peers. Default no (authenticate all connections).

Dtmfmode

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

Trustrpid

Whether Asterisk should trust the RPID settings from this device. Usually should be yes for CONNECTEDLINE( ) functionality to work if supported by the endpoint.

Sendrpid

Whether Asterisk should send RPID (or PAI) info to the device. Usually should be enabled to the settings used by your device for CONNECTEDLINE( ) functionality to work if supported by the endpoint.

Prack

The PRACK request plays the same role as ACK, but for provisional responses.

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 messages 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

Avpf

Whether to Enable AVPF. Defaults to no. the WebRTC standard has selected AVPF as the audio video profile to use for media streams. This is not the default profile in use by Asterisk. As a result the following must enabled to use WebRTC.

Icesupport

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

Dtlsenable

Whether to Enable DTLS for this peer. Defaults to no.

Dtlsverify

Whether to verify that the provided peer cerificate is valid. Defaults to no.

Dtlssetup

Behavior on DTLS incoming and outgoing connections. Defaults to actpass.

Dtlscertfile

Path to certificate file to present

Dtlscafile

Path to cafile file to present

Dtlsprivatekey

Path to private key for certificate file.

Encryption

Whether to offer SRTP encrypted media (and only SRTP encrypted media) on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if the peer does not support SRTP. Defaults to no.

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.

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

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

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 

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 '

Item

Definition

No Routes Available 

Standard Routes

Message or tone to be played if no trunks are available.

Intra-Company Routes

Message or tone to be played if no trunks are available. Used on routes marked as intra-company only.

Emergency Routes

Message or tone to be played if no trunks are available. Used on all emergency routes. Consider a message instructing callers to find an alternative means of calling emergency services such as a cell phone or alarm system panel.

Trunk Failures

No Answer

Message or tone to be played if there was no answer. Default message is:"The number is not answering." Hangupcause is 18 or 19

Number or Address Incomplete

Message or tone to be played if trunk reports Number or Address Incomplete. Usually this means that the number you have dialed is to short. Default message is:"The number you have dialed is not in service. Please check the number and try again."Hangupcause is 28

Options

Definition

Country

Configuration for location specific tone indications.

Dial tone

Set of tones to be played when one picks up the hook.

Busy tone

Set of tones played when the receiving end is busy.

Congestion tone

Set of tones played when there is some congestion.

Record tone

Set of tones played when call recording is in progress.

Ring cadence

List of durations the physical bell rings.

Ring tone

Set of tones to be played when the receiving end is ringing.

Call waiting tone

Set of tones played when there is a call waiting in the background.

Dial recall tone

Many phone systems play a recall dial tone after hook flash.

Info tone

Set of tones played with special information messages (e.g., number is out of service.)

Stutter tone 

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.

Definition

Jitter Buffer Settings

Jitter Buffer

Asterisk: jbenable. 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 can not 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. An example is if receiving from a jittery channel to voicemail, the jitter buffer will be used if enabled. However, it will not be used when sending to a SIP endpoint since they usually have their own jitter buffers. See jbforce to force it

Advanced General Settings

Language

Default Language for a channel, Asterisk: language

Default Context

Asterisk: context. Default context for incoming calls if not specified. FreePBX sets this to from-sip-external which is used in conjunction with the Allow Anonymous SIP calls. If you change this you will effect that behavior. It is recommended to leave this blank.

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. It is recommended to leave this blank. Asterisk 1.8 all supports IPv6. An address of '::' will listen on both ipv4 and ipv6.

Bind Port

Asterisk: bindport. Local incoming UDP Port that Asterisk will bind to and listen for SIP messages. The SIP standard is 5060 and in most cases this is what you want. It is recommended to leave this blank.

Allow SIP Guests

Allow Anonymous Inbound SIP Calls

SRV Lookup

Enable Asterisk srvlookup. See current version of Asterisk for limitations on SRV functionality.

Call Events

Generate manager events when sip ua performs events (e.g. hold).

Other SIP Settings

You may set any other SIP settings not present here that are allowed to be configured in the General section of sip.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

Registration Settings 

Registration Times

Asterisk: minregexpire, maxregexpire. Minimum and maximum length of time that IAX peers can request as a registration expiration interval (in seconds).

Jitter Buffer Settings 

Jitter Buffer