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.
Sincerely,
Brian LaVallee
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