[Asterisk-Users] SIP Phones-Receptionist Setup

Simon Brown Simon.Brown at otterson.com.au
Tue Nov 23 00:37:27 MST 2004


Have you looked at FOP - available at www.asternic.org
It might be most or part of the way to what you want.  You could work with
Nicolas on adding the features that you need and it doesn't have.

Just a thought.

Simon 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Blackham
Sent: Tuesday, 23 November 2004 18:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phones-Receptionist Setup

I have a 200 and the hint() stuff works fine for indicating status of any
channel (including Agent channels).  The Snom subscribes to asterisk at
whatever url you put in there, then * will send notify events when the dialog
state changes.  It's not quite a shared-line (at least the way I understand
it) but it does dial the extension and show status.  Adding the extended
keypad to a 220 is just 'more buttons' that are all configurable the same,
unlike the Cisco keypad which doesn't do SIP.

I'm also working on a receptionist panel that I intend to operate with a
touchscreen LCD (probably a 15", there are plenty that have X support).
It'll not only subscribe to dialog state of asterisk channels, but will also
subscribe to SIP presence (Polycom phones in our case), so she knows if
they're on DND, or whatever they set their status to.  On the back end,
there's an XMLRPC daemon that tracks all this state and abstracts the dirty
work of transfers, etc, via the manager port.  It also will be doing a number
of other things for our in-house Java call center apps.

I chose to go this route instead of just the Snom since she would rather have
it on a PC anyway (much easier to find your target, can change to suit her),
and mainly since DND on SIP phones simply bounces the call as busy without *
being able to tell the receptionist not to bother in the first place.  Also,
transfers will work better this way since it's going to be a native * method,
and not SIP refer, with options for vmail busy, vmail unavail, blind,
attended or camp-on (my dial plan is heavily tweaked).

I plan to have this posted up under GPL by February.  It'll be in
C++/Qt or Java.


On Sat, 20 Nov 2004 16:45:27 -0500, Curren C. Calhoun
<asterisk at currencalhoun.com> wrote:
>  Fred is in sales... A call comes into the receptionist and they 
> transfer the call to Fred.  The receptionist can tell Fred is still on 
> the phone by viewing the assigned key on the Snom 220's keypad, so if 
> another call comes in they know he is on the phone instead of just 
> blindly transferring the call and pushing the person to his voicemail.  
> So they can ask the person hold or if them want to be transferred into
Fred's voicemail.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list