[asterisk-biz] General Asterisk Question

Tue Jul 4 18:06:23 MST 2006

> Just out of interest; what is your "financial company" environment?

  We are a small company servicing consumer and mortgage loans with a small
inbound center and collections group.

> Not until you commit the number to the network by pressing the send key do
you initiate a call.

  I understand this.  I am looking to provide a usability feature from the
old TDM world average users still expect.  They don't care about technology.
They simply look for it to work, and if it works different than they are
used to they either don't like it or devaluate it.  I don't agree but they
pay the bills and I look to provide them what they ask for with a minimum of
excuses.

> 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.

  I tried a Linksys.  It was ok, but I bailed on it quick since the model I
had did not do POE.  I also am not interested in dial plans inside phones as
I want to keep the call control and management in a common database.  I
belive dial plans belong in the main system and not at the device.  Managing
each phone's dial plan sounds like more work than its worth.  Reminds me of
the old "KSU-less" phone system offered 20 years ago.  Yeech!

> Why are you trying to wrestle with H323?  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?

  I use Asterisk as a gateway from out of state branch office and SOHO
deployments using IAX2.  The T1 allows a branch user to dial 4-digit Avaya
extensions and ring their phone.  I have routes in the Avaya to allow an
Avaya deskset to 4digit dial a branch or SOHO user at the end of an IAX
line.  The brach & SOHO dial 9 and originate LD our Avaya T1 pipes.  I've
also recenting added a new "internet" route to my Avaya ARS that goes out
Asterisk.  We are building a centralized single managed system regardless of
where the end user resides.  I'm not looking to blow away the Avaya, just
make both systems live together happy.  My reason for needing H.323 is that
I am breaking the T1 down coming out of the Avaya with a digital cross
connect and only have 18 channels pointed at Asterisk with the other 6
pointed at another system.  This T1 is RBS and I cannot pass inband ANI
across it to Asterisk from my S8300.  I'm hoping I can use H.323 between
Asterisk and the Avaya with the intent of replacing my Audix voice mail.  I
like Asterisk's voice mail a lot better but need to pass it ANI and Avaya
station numbers to have it work seamless like Audix does.

Thanks for the help.
Your suggestions are appreciated.
Jim

-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Mark Phillips
Sent: Tuesday, July 04, 2006 3:12 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] General Asterisk Question

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>

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