[asterisk-users] The plain old PBX functionality

Brian LaVallee b.lavallee at globaltank.jp
Fri Aug 8 02:29:33 CDT 2014

On 8/8/14, 14:05, Gergo Csibra wrote:
> Hi,
> back in the old analog telephony days there was "digital" PBX-es and
> digital "system" phonesets. This phonesets have had many individual
> illuminatable buttons connected with extensions. The PBX can show on
> the buttons if some extension is ringing (blinks) or busy (constant
> light), and the user can transfer the call with one touch (pressing
> one of this button).

Because of the peer-to-peer nature of SIP, many of the digital PBX 
features can be difficult to reproduce.

If you consider where the 'brains' of the system reside, you can see the 
reason.  In the traditional digital PBX, all functionality was 
controlled by the PBX itself.  It was a Master/Slave communication 
model.  Phones were basically dumb terminals, how a button functioned 
was determined by the digital PBX.

With SIP, phones and servers are peers.  Master/Slave roles don't exist 
with SIP.  Control is determined by the device that initiated the 
session.  I will not go into the pro's and con's.  But by dialing a URL, 
it's possible to entirely exclude 'the server' from a call.

> I search this functionality in Asterisk. What versions, and what
> extension functions (or other settings), and what VoIP phones can do
> this?
The key thing in the SIP architecture to understand, the server DOES NOT 
control the phone.  How a button functions depends on how each 
individual phone is configured.  How a phone reacts to an instruction, 
depends on how the phone is configured.

While it's possible to host a phone configuration template on the 
Asterisk server for all phones to use, it's actually independent from 
the Asterisk software.

Depending on the make/model of the phone, most of the basic features 
(hold, transfer, redial) are available by default.  To duplicate the 
digital PBX features you're looking for, will involve two groups of 
settings.  Configuration on the server -and- configuration on the phone.

SIP phones are NOT dumb terminals, you have to configure them to operate 
how you want.

Brian LaVallee

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