[asterisk-biz] General Asterisk Question

[Mark Phillips]

You're right, this is a question for the user list but let me try to
answer it.

Just out of interest; what is your "financial company" environment? I
have a 250 seat install working on Wall Street for a firm supplying a
managed OMS. We use their old Definity G3 to route calls to and from the
PSTN purely because they have invested in it and can't justify
abandoning it yet in terms of cost. Feeding the Lucent pig was what led
them down the Asterisk road.

The "problem" you are experiencing is the norm in the VoIP world. Not
until you commit the number to the network by pressing the send key do
you initiate a call. Your phone has no connection with the phone system
until that point (save some keep alive messages).

I *believe* that the only VoIP protocol that sends numbers as they are
dialed is Skinny. 

IMHO you made a bad choice with your GS2K's. If you had bought something
like a Cisco/Linksys/Sipura or even a Polycom you'd be able to have dial
plans loaded into the phone. They would do a similar function to that
which your Lucent is doing; they'd trap the dialed digits and then dial
upon a match without having to press the send button (# key on most
phones).

Why are you trying to wrestle with H323? Do you require IP-IP
connections with your Lucent handsets? The standard for Asterisk is SIP.
As you have a T1 interconnect from your Lucent to your Asterisk you are
forced to use the Asterisk server from beginning to end of your call.
Why not use SIP?

Mark

On Tue, 2006-07-04 at 10:00 -0500, Jim Houser wrote:
> Hi,
> 
>   Please accept my apologies in advance.  This question may be more suited
> to the user list, but I would have to believe people deploying Asterisk
> professionally have had to deal with this.  I manage an IT department in a
> financial company and am trying to integrate Asterisk into it beside an
> Avaya switch.
> 
>    I started playing with AAH and tried a few other GUIs.  Currently I have
> been happiest with the Pound Key build and doing everything manually.  I
> miss some of the GUI but have found this the most flexible for our needs.
> 
>   My question my be dumb but I just need to ask.  I've got past basic dial
> plans and adding features.  I currently have Asterisk networked with our
> Avaya S8300 via T1.  I am struggling with H323 but should get past it, (any
> hints are welcome as I can't find much regarding Pound Key).
> 
>   My reason for writing is there is one item I would like to improve upon
> but it may be something SIP based and not possible to change.  ???
> 
>   The standard "accepted and expected" operation of a PBX, (yes I'm an old
> telecom guy), is for the PBX to collect digits and when it has enough digits
> to fit into a route it selects it and outpulses.  From the end user they
> dial 9, dial tone is not broke as the 9 is just an access code as the PBX is
> waiting for digits, then upon the next digit dial tone is broke digits are
> collected and it dials out when the dialed number fits a route.  Due to the
> route patterns if it fits in 7 digits the dial starts immediately after the
> 7th digit, you already know this...
> 
>   On Asterisk, to call out you dial and press send, (for example on my
> Grandstream 2000s - I can't get my Avaya 46XX phones to stay registered on
> Asterisk).  My users see this as "cell phone" operation and somehow that
> lowers their perceived value of Asterisk.  I know, stupid, but it is what it
> is.  Has anyone built a dial plan that emulates the original PBX operation
> at the deskset removing the need to push a send button at the phone?
> 
> Thanks, in advance, for any feedback.
> Jim
> 
> 
> 
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-- 
Mark Phillips <g7ltt at g7ltt.com>