asterisk users - Mar 2006 - Flash zap trunk from Softphone or IP Handsets...

Hi Guys!!

 

I wrote a little patch for asterisk 1.2.5 and I will maintain it for future
release unless somebody explains me how we can ask people at Digium to add it to
the source tree...

 

We are planning on using Asterisk as our main PBX for the office over the next
few weeks.  Our current setup uses TDM400 cards to bring our 8 lines into
Asterisk, our Telco provides us an option for three way calling and call
transfer on the lines, option that we use extensively, we rarely do conf call
between people that are in the office and often somebody will conf me on my cell
from an office line and so on. Anyway to make a long story short we needed an
option to flash the zap trunk from either a soft phone or an IP handset.  I
looked at all the available workaround like transferring the call to an
extension that would flash and transfer the call back to me, or I found on
voip-info.org somebody using the features.conf to make an option such as *3 to
flash, but the limitation with that procedure is you have to know who placed the
call to use the correct feature call (like *3 would flash the caller and *4
would flash the callee, depending on whether you received the call or made it)
which I found some what crazy.  So I added a detect context to features.conf
which then finds which channel is the zap one and call the flash on the right
channel for you.

 

So here what I have.

 

A patch that works on asterisk 1.2.5.

 

The patch is available here
http://mail.guardianmicro.com/asterisk/map_flash.patch

(I put it up on my website just because I'm curious to see how many will
even care of downloading it :-) )

 

My features.conf contain something like:

zapflash => **,detect,Flash,()

 

My Extensions.conf looks like this:

[globals]

DYNAMIC_FEATURES=>zapflash

 

[guardian]

[...]

exten => _9NXXNXXXXXX,1,Dial(Zap/1/${EXTEN:1},,wW)

 

[incoming]

exten => s,1,Answer()

exten => s,n,Wait(2)

exten => s,n,Dial(SIP/221,,wW)

exten => s,n,Playback(pls-wait-connect-call)

exten => s,n,Flash()

exten => s,n,SendDTMF(mycellphonenumberhere)

exten => s,n,Wait(5)

exten => s,n,Hangup()

[...]

 

So any of my users can press ** to flash the line from any SIP/IAX or else
technology, the only pit fall would be somebody flashing from another Zap
extension, but I could fix that as well if any body cares of such scenario...

 

Anyway if you plan on using it just drop me a line I would like to know about
your experience with it.

 

Thanks!! Long life to Asterisk!! 

 

          Marc-Andr? Poupier, MCSE

            Chief System Architect 

        Guardian Microsystems Inc. 
            The Invisible IT People ? 
     Tel: 514.788.1108  Fax: 514.684.8781

        mapoupier@guardianmicro.com 

 

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Hi Marc-Andr?,

You should probably post this to the asterisk-dev mailing list. That's
where people dealing with the source code talk and listen. The list is
also MUCH lower volume thank goodness!

Mark.

On 3/13/06, Marc-andre Poupier <mapoupier@guardianmicro.com>
wrote:
> Hi Guys!!
>
> I wrote a little patch for asterisk 1.2.5 and I will maintain it for future
release unless somebody explains me how we can ask people at Digium to add it to
the source tree?
>
> <snip>