Paging and Intercom

On legacy phone systems you can find the following kinds of paging:

• Dial a code to connect to a separate overhead paging and announcement system (like in an airport)
• Dial a code and connect directly to a built-in one-way announcement speaker on one or more phones
• Dial a code and connect directly to a built-in two-way announcement and talkback function on one or more phones

Some overhead paging systems also provide a talkback system so that the person being paged can just speak to respond. Background noise issues limit where this feature can be used. The talkback function is usually setup to be hands free. That means that the person responding to the page does not need to take any action other then speaking.

New in Asterisk 1.2: The new dialplan command Page utilizes MeetMe to page one or more phones.

New in Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is:

MulticastRTP/<type>/<destination>/<control address>

Type can be either basic or linksys. Destination is the IP address and port for the RTP packets. Control address is specific to the linksys type and is used for sending the control packets unique to them.

Advanced Paging / Intercom

There is also another system available since many years, the best one, combining paging and intercom. Here the talback system is limited to only one phone. The paging is done in one way mode through a group of phones, and the person being paged can respond pressing a digit to switch the nearer phone to two-way mode, simultaneously hanging-up all other phones speakers.
This mode combine the best of the two world, eliminate the noise problems, and keep the communication private as soon as the paged person pressed the right digit on a phone.
It should be possible to implement this mode on Asterisk with a managed conference and a feature map application.

Multicasting begin to be supported at all major phones manufacturers, Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast listening address. This will permit to reduce the generated trafic for an extensive paging.

If a phone is in use when a page arrives, some systems can do a "whisper page" so that only the person being paged can hear the page.

SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. The phones most often mentioned supporting this are:

• Digital Acoustics Intercoms and VoIP Paging Products
• Aastra/Sayson480i, 9133i, 480i CT Cordless 5xi phones supported with firmware update March 2008 - Configuration Instructions: Aastra/Sayson AutoAnswer
• Cisco 7940/7960 - see: Cisco 7940-7960 auto-answer config
• Polycom 600 - see: Polycom auto-answer config
• Grandstream GXP2000 and BudgeTone phone supports Auto Answer with firmware 1.0.4.54 or later. ...