PBX feature summary

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This section provides a general overview of various PBX features available in PortaSwitch, as well as their activation and usage. Please note that many of these features are either handled entirely on the IP phone or require adequate support from it; such cases will be clearly indicated in the feature descriptions. Also, for your convenience, we have provided instructions about how particular features can be used on an IP phone. These instructions are applicable to Sipura/Linksys devices (1000, 2000, 2100, and 3000). For other types of IP phones, please consult the manual provided by the vendor.

Additional authorization / authorization codes for toll calls

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Feature description: This feature allows you to perform additional verification of outgoing tolls on international calls. Especially in the case of a single phone being shared among multiple users, this feature enables individual user accountability.

Supported by PortaSwitch. See the Additional Authorization for Toll Calls handbook for more details.

Alternate numbers

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Feature description: In addition to a user’s main phone number, multiple alternate phone numbers can be assigned, all of which will ring on that user’s IP phone.

This is implemented by assigning additional aliases to the account representing the main phone line. Each alias is basically a direct inward dialing (DID) number.

Anonymous call rejection

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Feature description: Automatically reject incoming calls from parties who do not deliver their name or telephone number with the call.

Make sure your IP phone supports this feature (e.g., Sipura). To activate it, dial the *77 code; to deactivate it, dial the *87 code.

Auto attendant

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Feature description: This provides IVR for callers and allows them to navigate among different options by pressing phone keys. Auto attendant capabilities include simple features such as playing a certain voice prompt to an end user or collecting his DTMF input, as well as more advanced features such as detecting incoming faxes or call queues.

See the Auto Attendant chapter for more details.

Automatic line / direct connect (“Hotline”)

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Feature description: Automatically dials a pre-assigned PBX station’s extension number or external telephone number whenever a user goes off-hook or lifts the handset.

This feature is configured on the SIP phone using the dial-plan configuration parameter. For example, the following implements a Hotline phone that automatically calls 1 212 5551234: ( S0 <:12125551234> )

The following creates a warmline to a local office operator (1000) after five seconds, unless a 4-digit extension is dialed by the user: ( P5 <:1000> | xxxx )

Busy Lamp Field (BLF)

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Feature description: The Busy lamp field (BLF) feature monitors statuses of individual phone lines (idle, busy, etc.) within the same PBX environment and displays them in real-time on the attendant phone console (IP phone with BLF).

This feature is implemented in the presence server. The only thing required from the endpoint is to subscribe to notifications regarding particular phone lines.

Call recordings

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Feature description: It shows the most recent calls and call details to an end user. It also provides the ability to download the recorded calls (if any were recorded) or delete them.

Supported by PortaSwitch via the Dashboard feature on the account self-care interface.

Call forking / simultaneous ringing

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Feature description: Allow all SIP phones registered on a single account to ring simultaneously. Consequently, if an end user owns three SIP phones (e.g., a mobile application on a smartphone, a tablet and a desktop IP phone), they can receive calls to all three devices simultaneously. The same account ID and password can be applied for all end user SIP phones.

See the Call forking chapter for more details.

Call forwarding

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Call forwarding always

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Feature description: Automatically routes all incoming calls for a given extension to another number (extension, home/mobile phone, etc.).

This feature is implemented by provisioning the call forwarding/follow-me service and setting the Default Answering Mode to “Forward Only.”

Call forwarding when busy

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Feature description: Automatically routes incoming calls for a given extension to another pre-selected number when the first extension is busy.

This feature is implemented by provisioning the follow-me service and activating the Cfwd Busy Serv supplementary service on the IP phone. Use the *90 code to activate this feature, and code *91 to deactivate it.

Call forwarding to voice mail always

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Feature description: Automatically routes all incoming calls for a given extension to voice mail.

This feature is implemented by setting the Default Answering Mode to “Voicemail Only.”

Call forwarding to voice mail when busy

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Feature description: Automatically routes incoming calls to voice mail for a given extension when that extension is busy.

This feature is implemented by setting the Default Answering Mode to “Ring then Voicemail” and then disabling Call Waiting.

Call forwarding to voice mail when call unanswered

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Feature description: Automatically routes incoming calls for a given extension to voice mail after a specified number of rings when there is no answer.

This feature is implemented by setting the Default Answering Mode to “Ring then Voicemail.”

Call forwarding on don’t answer

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Feature description: Automatically routes incoming calls for a given extension to another pre-selected number when there is no answer after a specified number of rings.

This feature is implemented by provisioning the follow-me service (choose “Follow-me when unavailable,” then set the ring timeout parameter in follow-me). You may also utilize this feature on the IP phone itself by activating the Cfwd No Ans Serv supplementary service. Use the *92 code to activate this feature, and *93 to deactivate it.

Multi-path forwarding

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Feature description: Indicates the number of forwarded calls (originally dialed to the same PBX extension) that may occur simultaneously.

This feature may be implemented similarly to other call forwarding scenarios, only in this case the follow-me service should be provisioned with a simultaneous ring option.

Phone-initiated forwarding

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Feature description: The phone can be programmed with a “forward to” phone number and subsequent incoming call requests will be answered by “302” responses.

This feature may be implemented similarly to other call forwarding scenarios, but advanced settings such as multiple forwarding numbers, simultaneous ringing and time periods will not be available for phone-initiated forwarding.

More detailed information about this feature can be found in the Call forwarding from an IP Phone (Endpoint redirection) chapter.

Call me now

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Feature description: This allows an end user to request a call to the user’s phone from the service provider’s helpdesk (at the expense of the service provider).

See the Web Callback Trigger chapter in the PortaSIP Media Applications Guide for more details.

Call parking

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Feature description: This allows a user to place a call on hold, move to a different location, and then resume the call from any other station within the cloud PBX by dialing a retrieval code.

Supported by PortaSwitch. To use this feature, a customer should define a “call parking prefix” in their call features configuration. Then, while a phone conversation is taking place, the user can simply place the call on hold and dial a specific call parking prefix. Then they will hear the dynamically assigned “retrieval code.” This retrieval code can be dialed from any phone of PBX customer to retrieve the conversation (i.e., connect the call to that phone). It is also possible to quickly retrieve a call from an original phone by dialing a special “release prefix.” See the Call parking chapter for more details.

Call queues

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Feature description: This provides “call center” functionality. When a large number of incoming calls arrive to the auto attendant from customers, PortaSwitch can forward these calls to the actual agents (customer service representatives) in a regulated fashion.

See the Call queues chapter for details.

Call recording

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Feature description: This allows a user to record all incoming/outgoing /redirected calls so they can be listened to (or downloaded) from the self-care web portal later on.

Supported by PortaSwitch via the Call Recording feature.

On-demand call recording

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Feature description: This allows an end user to start/stop call recording at any time during a call.

Supported by PortaSwitch via the Call Recording feature.

To enable a specific account to start/stop call recording at any time during a call, the administrator enables the Allow to start/stop recording manually option in the call recording configuration.

There are two approaches to activating the on-demand call recording:

  1. By pressing the “Record” button on the IP phone. This approach enables end users with particular IP phone models to activate the on-demand call recording at the push of one button. The main requirement for an IP phone is to send a special SIP INFO request with the header “Record” with “On” and “Off” content. Then you need to assign the call recording to a specific button on your IP phone (for example, Yealink SIP-T28P).
  2. By dialing DTMF codes. This approach is available for all end users, regardless of the IP phone model they use. To start/stop the recording, the end user dials DTMF codes. The DTMF codes are specified in the dialing rule of PBX type. For example, the end user dials *44 to start recording during a call and *45 to stop the recording.

Call recording announcement

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Feature description: Businesses can comply with call recording regulations that require everyone on a call to be notified if the conversation is being recorded. As soon as recording begins, a call recording announcement automatically plays. The recorded call always contains the announcement, ensuring that the call parties are notified of the call recording.

See the Call recording announcement chapter for details.

Call return

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Feature description: This allows the user to call the last party or number that called the user, regardless of whether the user answered the original call or knows the caller’s identity.

The feature is provided by the IP phone. Dial the *69 code to use this feature.

Call supervision

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Feature description: PBX customers can supervise their colleagues’ calls in real time. For example, a manager can listen in on a sales agent’s calls, guide them, and, if needed, participate in the call.

The manager can choose any of three modes when joining an active call and switch between modes during the call:

  • Spy mode – the agent and the client are not aware of the manager’s presence;
  • Whisper mode – only the agent can hear the manager; and
  • Barge-in mode – both the agent and the client can hear the manager.

See the Call supervision chapter for details.

Call trace

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Feature description: This permits end users to request a trace on unwelcome calls if they want to initiate an official investigation.

See the Tracing unwelcome calls chapter for details.

Call transfer

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Feature description: Transfers an existing call to another party (inside or outside the PBX group).

PortaSwitch supports the following transfer types:

  • Unattended (blind) transfer
  • Attended/supervised transfer (transfer with third-party consultation)
  • Attended transfer of forwarded calls by DTMF

See the Call Transfer chapter for more details.

Call waiting

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Feature description: A feature that allows users to be alerted of one or more calls awaiting connection during a current conversation. Users are normally notified by a short tone on the phone or by use of the caller ID feature. Then, they can answer the second call, while the first one is still on hold.

Calling line ID delivery

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Feature description: Allows the user to identify the name and telephone number of a calling party before answering an incoming call.

Supported by PortaSwitch. The phone must have a display to show the caller ID.

Calling line ID on call waiting

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Feature description: Allows a caller’s name and number to be displayed when the called party is taking another call.

Supported by PortaSwitch. The phone must have a display to show the caller ID, and the Call Waiting feature must be activated.

Calling group ID delivery

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Feature description: This allows the user to identify the name and number of a hunt group that the calling party belongs to before answering an incoming call. This allows you to fine-tune the identity to be used for calls made by separate departments in your company.

Supported by PortaSwitch.

Calling line ID blocking

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Feature description: This allows an end user to indicate that he wants privacy for a particular outgoing call, i.e., the other party will not see his phone number.

This can be done by either activating the privacy settings on the IP phone itself (in this case, the IP phone will include the corresponding RPID header of the SIP INVITE), or by activating the Hide CLI feature on the PortaSwitch side. See the Support for privacy flags chapter for more details.

Calling name retrieval

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Feature description: This allows an end user to see a caller’s ID (name and surname, or company name that owns the number) in addition to the original caller’s number. See the Caller ID (CNAM) lookup chapter for more details.

Calling plan – forwarded/transferred

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This plan enables administrators to prevent users from forwarding or transferring calls to certain types of numbers, such as long distance, toll, or premium numbers.

Supported by PortaSwitch. Forbid required destinations in the tariff and define it within the product configuration with the FOLLOWME access code.

Calling plan – incoming

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Feature description: This allows the administrator to define a set of rules that will be applied to every incoming call (e.g., to prevent users from receiving calls from outside the company, or forward calls from certain destinations directly to voicemail).

See the Call screening chapter for details.

Calling plan – outgoing

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Feature description: This allows the administrator to prohibit outgoing calls, such as long distance, toll, or premium calls, made by individual users to specific destinations.

See the Call Barring section for details.

Communication barring

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Feature description: Prevents certain types of calls from being made or received by particular stations.

For example, phones in public areas can be blocked from originating calls to external numbers, so as to prevent unauthorized users from incurring toll charges. Phones in certain areas may be blocked from receiving external calls in order to limit employees’ ability to take personal calls. A wide variety of restrictions are available, covering incoming calls, outgoing calls, toll restrictions, code restrictions, and differential treatment for internal and external calls.

Provided via the tariff configuration in PortaBilling or by using the Call barring feature.

Conferencing

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Feature description: Allows an end user to create and manage conference bridges for instant meetings.

See the Conferencing chapter in the PortaSIP Media Applications Guide for more details.

Control call waiting

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Feature description: Enables/disables delivery of the call waiting feature to IP phones, allowing administrators to control call waiting for a specific account. This ensures that the feature is supplied only to users who have it activated on the PortaSwitch side (regardless of whether it is enabled on the IP phone itself).

Supported by PortaSwitch.

Configurable time zones

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Feature description: This allows an administrator to define specific time zones for every PortaSwitch user (customer, reseller, vendor, etc.). The respective time zone is used for services that require date/time stamps, such as Messaging, Auto attendant. When a user logs in to their web interface, all date and time information will be shown in the user’s specified format.

See the Date and Time Information chapter of the PortaBilling Administrator guide.

Dialing rules / PBX dialing transparency

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Feature description: This allows an administrator or a customer to define a way of dialing phone numbers that is convenient for end users.

Supported by PortaSwitch.

The dialing rule wizard can be used to construct correct rules based on the parameters provided, such as country or area code. Alternatively, the dialing rules can be defined by means of regular expressions. This allows administrators to easily manage a network that has many different customer numbering plans.

 ANI translation on incoming / outgoing calls

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In addition, the dialing rules can translate the CLI (ANI) numbers to/from a vendor-specific format when routing a call to/from a vendor’s network. This enables the sending of caller information to a vendor in the format that he requires (e.g., a 10-digit phone number for US callers).

DID (Direct Inward Dialing Number)

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See Alternate numbers.

DOD (Direct Outward Dialing)

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Feature description: This enables a user to connect to outside lines directly, without the need to go through an operator or dial other numbers first.

Supported by PortaSwitch.

DISA/Remote office/two-stage dialing

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Feature description: Allows the user to log into PBX from the outside network and use PBX features as if from their own extension. During login, the user is asked to enter their DISA password. If the password is valid, they can use PBX services, such as dial local users, make out-group calls, etc.

Supported by PortaSwitch.

Directed group pickup

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Feature description: Allows phones in the same PBX environment (all accounts under the same customer) to answer each other’s calls by dialing a Group pickup prefix on their phones.

Supported by PortaSwitch.

Distinctive ringing

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Feature description: Uses a special ringing pattern to indicate whether an incoming call is from inside or outside the PBX group.

Supported by PortaSwitch for the Ext-to-ext call distinctive ring feature.

Diversion inhibitor

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Feature description: This allows an administrator to override voicemail settings of a particular extension that belong to a hunt group. Thus, when a call is made to the hunt group, it is not redirected to voicemail on a specific extension but rings on this hunt group’s other extensions.

Supported by PortaSwitch. Enable the Ignore follow-me/voicemail option when configuring a hunt group.

Do not disturb

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Feature description: The Do not disturb (DND) feature allows end users to temporarily disable incoming calls.

Supported by PortaSwitch. A SIP phone is required to support the DND feature.

Enterprise-wide directory

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This allows administrators to organize commonly used phone numbers under a general directory and enables users to dial them using short numbers.

Supported by PortaSwitch. Configured on the customer self-care portal. Specify the maximum number of digits a short number can consist of and define a list of phone extensions on the Abbreviated Dialing tab.

Extension dialing

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Feature description: This allows an end user to dial extension numbers for quickly connecting with phones inside of the same PBX environment.

Supported by PortaSwitch

Flash call hold

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Feature description: Calls can be put on hold by depressing the switch-hook or pressing the flash button. After completing the second call, the user is automatically reconnected to the original call on hold.

Supported by PortaSwitch.

Group calling line identity

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Feature description: This service allows a user identity (name and number) to be defined for a group of users.

See the SIP identity chapter for more details.

HD voice

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Feature description: This provides better audio quality during calls. Both user and vendor equipment must support wideband codecs (e.g., G.722).

Supported by PortaSwitch. Configure the codec policy for accounts and for connections.

Hunt groups

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Feature description: This allows calls to be redirected to other predetermined lines when the line called is busy. Hunting allows a number of lines to be grouped into a “pool,” so that incoming calls are directed to whichever of these lines is available.

The following hunt sequence modes are supported:

  • Circular (Order) – extensions will be called one by one from the first (topmost) to the last number until the call is answered.
  • Simultaneous – All extensions in the group will ring simultaneously.
  • Random – the call will be delivered to extensions in random order.
  • Least used – this sorts the phone lines in descending order beginning with their last usage, and delivers a call to their extensions, accordingly.

Supported by PortaSwitch. Hunt groups are defined on the Hunt groups tab on the customer self-care portal.

Stop/resume receiving hunt group calls

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Feature description: Hunt group members can control whether to receive their hunt group calls. To stop receiving calls, they can simply log out of the hunt group – and log back in as soon as they want to resume. When logged-out, the hunt group members can still receive direct calls.

See the Stop/resume receiving hunt group calls chapter for more details.

IP device/phone inventory

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Feature description: The IP phone directory allows you to keep track of IP devices (SIP phones or adaptors) that are distributed among your customers.

See the CPE inventory chapter.

Last number redial

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Feature description: Enables users to redial the last number they called by clicking the Redial button.

Provided by an IP phone. Supported by PortaSwitch.

Message waiting, audible

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Feature description: This provides the user with an audible notification – a “stutter” dial tone when messages have been left in the extension’s voice mail system.

Provided by the IP phone and supported by PortaSwitch (the actual “message waiting” SIP info packet is originated by the Media Server and relayed by PortaSIP).

Message waiting visual

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Feature description: This provides the user with a visual indication when messages have been left in the company’s voice mail system.

Supported by PortaSwitch (the actual “message waiting” SIP info packet is originated by the Media Server and relayed by the Switching Server), requires the phone to be able to display the appropriate icon.

Modes to handle incoming calls (business hours, non-working hours)

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Feature description: PBX users can now use modes to control when and how calls come through to their line, and quickly change the way incoming calls are handled. For example, they can set their mode to accept calls during business hours, and to send all calls to voicemail at once during off-site meetings in order not to disturb colleagues in the office. Users can quickly change these settings by calling to the IVR or via their self-care interface.

See the Modes to handle incoming calls (business hours, non-working hours) chapter for details.

Multiple call appearances

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Feature description: Multiple call appearances allow each station to have two or more appearances of the user’s primary phone number. Each appearance gives the user the ability to handle one call. Consequently, Multiple call appearances allow the user to originate and/or terminate multiple calls simultaneously. Unlike an analog multi-line phone, the station needs only one line (and one phone number) for Multiple call appearances. When the user is involved in a call on one call appearance and another call is offered on a different call appearance, the user may use the Caller ID information to decide whether to answer the ringing call appearance or let the call be forwarded to voicemail. To answer the ringing call appearance (or originate a second simultaneous call), the user simply puts the first call appearance on hold. Calls on different appearances can be combined together to form a three-way conference call.

Supported by PortaSwitch via the Follow-me feature. The primary phone number (account) is provisioned on the IP phone, and all the other appearances are created as accounts with the follow-me configured to the primary account.

Multiple pickup groups

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Feature description: Allows phone lines in the same PBX environment to be grouped so that phone line owners within the group may answer each other’s calls by merely dialing a group call pickup prefix on their phones.

See the Multiple pickup groups chapter for details.

Music-on-hold

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Feature description: Provides a musical interlude for callers who are waiting on hold.

Supported by PortaSwitch. Every PBX user can upload their own melody or use the default one for their PBX environment.

MWI delivery to an endpoint

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Feature description: This feature allows the Media Server to automatically manage the SIP phone’s MWI status so that a user is notified when he has new messages.

Supported by PortaSwitch. A SIP phone is required to support the message waiting indicator (MWI).

Paging/Intercom calls (push to talk)

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Intercom calls

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Feature description: Allows a user to dial the other extension so that the destination IP phone automatically answers the call and activates the speakerphone mode. The user can start talking immediately, without waiting for the other handset to be picked up.

Group paging

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Feature description: Allows a user to use regular IP phones to broadcast announcements to members of a specific hunt group. All IP phones simultaneously broadcast the announcement via speakers. This is a one-way announcement, so users can’t reply to the caller.

Supported by PortaSwitch. A SIP phone must support paging/intercom features. See the Paging/Intercom calls chapter for more details.

Personalized name recording

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Feature description: This allows users to record and/or upload their greetings to be played back to callers (e.g., Voicemail greeting or Auto attendant prompts).

Supported by PortaSwitch.

Point-to-point video calls

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Feature description: This enables users to make video calls.

Supported by PortaSwitch and provided via phones that support video calls.

Privacy service

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Feature description: This enables users to exclude themselves or certain extensions from being accessible via the dial-by-name directory.

Supported by PortaSwitch.

Selective call acceptance

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Selective call acceptance (SCA) is a telecommunications system feature that allows customers to create a list of phone numbers from which they are willing to accept calls.

Supported by PortaSwitch via the Call screening module. Every PBX user can create rules defining a set of phone numbers. If an incoming call matches one of these numbers, the call is accepted; otherwise the call is rejected.

Selective call forwarding

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Selective call forwarding (SCF) is a telecommunications system feature that allows customers to forward callers from a selected group of numbers to another number.

Supported by PortaSwitch via the Call screening module. Every PBX user can create rules defining a set of phone numbers. If an incoming call matches one of these numbers, the call is forwarded to the destination defined in the call forwarding or follow-me settings.

Selective call rejection

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Feature description: Selective call rejection (SCR) is a telecommunications system feature that allows customers to reject incoming calls.

Supported by PortaSwitch via the Call screening module. Every PBX user can create rules defining a set of phone numbers. If an incoming call matches one of these numbers, the call is rejected.

Shared line appearance

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Feature description: Multiple IP phones can share a single phone number so that different PBX users can co-manage calls to/from that number. A user can answer a call from any IP phone. The call placed on hold can be resumed from another phone or from the same phone only (private hold). PBX users can see the current state (appearance) of the shared line, e.g., idle, busy, on hold, on any IP phone. Shared line can be programmed on multiple line keys on an IP phone, so users can handle multiple calls at once. Also, a user can join an active call from any other IP phone (barge into a call).

See the Shared line appearance chapter for details.

Speed dialing

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Feature description: This allows a user to dial frequently called telephone numbers using an abbreviated speed calling code instead of the entire number.

Supported by PortaSwitch via the Phone Book feature.

Station Message Detail Recording (SMDR)

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Feature description: Allows the corporate telecom manager to receive call detail records on a per-station basis before the monthly telephone bill is even issued. SMDR helps the customer control telephone fraud and abuse, perform accurate cost accounting, and analyze call patterns to identify opportunities for cost reductions.

Supported by PortaSwitch. Call details are available on the PortaBilling web interface.

Three-way conferencing (three-way calling)

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Feature description: This allows a user to add a third party to an existing conversation thereby forming a three-way conference call.

Supported by PortaSwitch. SIP phone must support the 3-way calling feature.

Toll restriction

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Feature description: Blocks a station from placing calls to telephone numbers that would incur toll charges.

Provided via the tariff configuration in PortaBilling or by using the Call barring feature.

Voice portal/self-care IVR

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Feature description: This enables users to access and manage personal settings (e.g., password setup, call forwarding options, greetings management, etc.) via any phone device.

Supported by PortaSwitch.

700/900 blocking

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Feature description: This blocks a station from placing calls to 700 and 900 numbers.

Provided via the tariff configuration in PortaBilling or by using the Call barring feature.

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