It's very possible that the Polycom IP600 will work with this.  As it is
just an implementation of a SIP standard for subscribing to the state of
other extensions.

As for the feature improvements you might see them from me, but not very
likely.  It is easier for me to train my customers to hit *8 (I will
probably just program a pickup button for them) than it is for me to
figure out what I have to do in code to accomplish a call pickup.

The conference stuff already works satisfactorily.  If a person is on
the phone you see their button lit, if you hit the button it calls them.
They hit ok to accept your call and their existing call goes on hold.
If they wish to conference they can this hit their conference button to
bridge the three of you together.   This is purely a function of the
phone.

More complex conferences I will achieve with use of the conference
application and the flash control panel.

You might, however, see the call parking bounty fulfilled by me when I
get the time.

David Hinkle

-----Original Message-----
From: John Todd [mailto:[EMAIL PROTECTED] 
Sent: Monday, August 30, 2004 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Snom Programmable button Mini Howto and
ringstate patch

At 1:23 PM -0500 on 8/30/04, David Hinkle wrote:
>The snom 200 and 220 have five programmable buttons.  Each button has a
>led that can be used to indecate if an extension is idle, in use, or
>ringing.  A button pannel for the 220 is also comming out soon that
will
>have 20'ish programmable buttons on board. 
>
>This is a killer app for any company that has receptionists handle
>calls, and pretty usefull for everyone else. 
>
>As a matter of fact, Asterisk already supports phone idle/in use states
>for the buttons, and at the bottom of this message you will find a
patch
>to enable the ring state.
>
>Howto:
>
>1. Configure the programable buttons as "destination" and enter the
>extension in the field.  After saving the page the phone will convert
>the extension to a sip url, which is fine.
>
>2. Modify your asterisk dialplan to provide "hints" that map a given
>extension to a channel.  (In asterisk, a channel can be busy or
ringing,
>but an extension is just a string of numbers that activate one or more
>applications).  Asterisk seems to provide syntax for allowing more than
>one channel to be mapped to any particular extension with the hint
>system, but I did not investigate that.
>
>Example:
>
>exten => 200,hint,SIP/RonC
>exten => 200,1,Macro(stdexten,SIP/RonC)
> 
>exten => 201,hint,SIP/JeanK
>exten => 201,1,Macro(stdexten,SIP/JeanK)
> 
>exten => 202,hint,SIP/JeffT
>exten => 202,1,Macro(stdexten,SIP/JeffT)
>
>3.  You must reload the dialplan and then reboot the phone for it's
>subscriptions to take effect.  After that, you should have working
>lights.
>
>4.  If you want the lights to blink on ringing, apply the following
>patch to the asterisk code. 
>
>You can not pick up a call by hitting the blinking button,  I was going
>to do this work but I decided to just train the receptionists to hit *8
>instead.   I have not studied this extensivly, but to implement it, i
>think it would just require asterisk to have support for sip "replaces"
>(I don't know if asterisk supports this or not) and the ringing notify
>needs to go out with a few more fields.  (It seems that the snom phone
>contacts the sip device listed in one of the ring notify message fields
>with an invite including a "replaces" header to pick up a call)
>
>I have also included a sip trace of a snom phone picking up a call
>placed to another phone using the blinking button in case anybody out
>there wants to tackle this problem themselves (Sample trace was
>collected when using snom phones with snom's sip proxy software).
>Please note that it seems like we must include the extra fields in the
>ring notify before the snom phone will procude the proper "replaces"
>invite in order to do a standards compliant call pickup.
>
>Notes on patch:
>If this patch is not in the proper format for submissions please
provide
>me a link to the asterisk submission policies.  It has been tested here
>at DerbyTech for about a week on our live phone system. 
>
>I submit this patch to the asterisk project under the GPL with hope
that
>it will be resubmited to CVS.
>
>Thankyou,
>David Hinkle
>Sr. Linux Engineer
>DerbyTech
>



This is pretty cool!  I might get a Snom phone just to try them out. 
You asked for comments, so here are a few:

1) Send the patch in "diff -u" format; that's the format used in the 
bugtracker.

2) You'll need to sign the disclaimer on the http://bugs.digium.com/ 
interface.  This disclaimer doesn't have much of a downside, and all 
patches to Asterisk for the public CVS have to be disclaimed in this 
way (avoids SCO-type lawsuits, etc.)

3) Have you looked at the configuration options for the Polycom IP600 
phones?  I don't know if this trick works with them, but they are 
pretty slick and have very programmable interfaces which may be 
almost compatible (or completely compatible) with this method.  I 
haven't looked, but that would be a very cool addition to those 
phones as well.

4) I'd say you've got 25% of the feature done.  Putting the extra 
effort into having the system pick up the call from any phone when 
one hits the flashing button would be I think another 25%.  Then, the 
final 50% would be if the button was pressed from a third-party phone 
while a call was already in progress that all three callers would be 
bridged together.  (more work than it seems, so I give it 50%.)  Bit 
by bit, Asterisk is getting there.

Asterisk in general needs to support more PBX-like features.  While 
it says it's an iPBX, it's still falling a bit short when compared to 
features found in even the most basic key system.  See my long posts 
over time on feature ideas that I've sent to -dev and -users.

JT
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