[asterisk-biz] proper analog behavior using an ATA

Sun Sep 28 16:50:21 CDT 2008

Alex,

  Please let your friend know I said "thanks" for his comments.  I agree with his thinking.

Jim

----- Original Message -----
From: "Alex Balashov" <abalashov at evaristesys.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz at lists.digium.com>
Sent: Sunday, September 28, 2008 4:42:11 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA

This opinion is offered by a friend of mine who saw the message, but who 
is not on the list.  Credit goes to Sean McCord (scm!!at!!cycoresys.com).

---

"The _only_ way to do analog right is with dedicated analog
hardware:  a channel bank.  You can "get it working" with ATAs and PCI
cards, but you will _always_ be fighting problems.  There will be plenty
of proponents of ATA and Digium cards for analog saying that they work
just fine.  There always are.  For a production system in which you will
actually be using features on the phone beyond the basic "connect a call
to the PSTN," and, more importantly, interfacing with other analog gear,
you will only drive yourself crazy and waste more money in time and
effort getting them to work.

Inevitably, the voltages are wrong, the ringing is insufficient, the
transmission levels are incorrect, the signalling isn't reliably passed,
timings are not what they should be, etc.  IMHO, these devices are made
by people and companies which are VoIP first and analog
"Johnny-come-lately"'s.  I have fought too many battles and have eaten
the purchase of too many channel banks when all else failed to ever
properly recommend ATA for analog PCI cards for most any professional
installation.

This is certainly true of Sipura/Linksys, Grandstream, Digium, and
Sangoma, by my own experience.  I am tentatively hopeful of the Rhino
analog PCI cards, but I only have one of those in the field on a system
which has light traffic and limited need of features.

Note that this distrust does not extend to a significant extent to
digital PCI cards.  I have generally not had nearly the problems (after
the first or second generation Digium cards, anyway) with them.  There
_are_ problems, of course, but they tend to be "hard" problems, not the
intermittant, degrading-over-time types of problems as with the analog
gear, so they can, at least, be dealt with.

Channel bank-based solutions _are_ more expensive on the hardware, and I
don't really see a way around that, but the trade-off is that, over
time, they will be _much_ _less_ expensive due to the lower maintenance
and troubleshooting costs.  I know it's a hard sell, but it is
definitely worth it.

As for alternatives to Adits, the CAC AccessBanks (I or II) can be
obtained cheaply on eBay and the like, if you get lucky (though watch
that the ABIs are not TR-08s, which will not work).  Rhino also makes
excellent channel banks, and if you are going to buy new equipment (as
opposed to used or eBay eq), they offer a good price, as well."

---

Jim Houser wrote:

> Hi all,
> 
>   Sorry, kinda long but please read...
> 
>   I'm looking for some help or correction if I'm overlooking something.  Let me preface this with I would be "the old guy on the block".  I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits.  Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software.  :-)
> 
>   I've dealt with most big name PBXs, Centrex, etc through the years.  I have a good data networking background and have a good grasp of common programming languages.  I have evolved with the industry, now I'm into VoIP and loving every minute of it.  I have been using Asterisk around 3 years.  Started with compiling my own and Asterisk at home.  
> 
>   Here's my issue I hope to get feedback and help with;
> 
>   I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones.  Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
> 
>   However, this is NOT the case with analog phones.  I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
> 
>   My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues.  The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone.  Basically a real feeling of cheap quality and "emulation" going on.
> 
>   In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times.  Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion).  Just throwing this out because it's another area where VoIP is behind the times.
> 
>   Now don't get me wrong.  I'm a major Asterisk evangelist and not pushing go back to TDM.  My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home.  That's where their expectation is.  What you sell better sound and work, in it's worst case, like the old TDM platforms did.  
> 
>   I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users.  That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards.  This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment. 
> 
>   What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service? 
> 
> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
> 
> Jim
> 
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-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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