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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