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.
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.
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
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.
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”)
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)
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.
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
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 always
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
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
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
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
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
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.
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.
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
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.
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.
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.
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
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:
- 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).
- 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
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.
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.
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.