[asterisk-biz] proper analog behavior using an ATA

[Steve Totaro]

Correction, bibliography = biography

On Sat, Sep 27, 2008 at 3:34 PM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

> See comments inline.
>
> On Sat, Sep 27, 2008 at 12:42 PM, Jim Houser <jhouser at trustamerifirst.com>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.  :-)
>>
>
> Why the bibliography?  I understand that you have been doing telephony for
> a long time.  So have I, but just not as long as you, same understanding
> though.
>
>
>>
>>  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.
>>
>
> Most "Old School Telephony Guys" Hate VoIP.  That may not be the case with
> you, but most people don't like change in general.
>
> Just for future reference, you don't really compile Asterisk at home, it is
> just an image that you install.
>
>
>>
>>  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.
>
>
> Give Quintum a try, they are excellent.  I have heard good things about
> Rhino and Xorcom.  Do you know they stopped using 25 pair lines for
> stations, that could be part of the problem ;-P
>
>
>>
>>
>>  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.
>
>
> To be fair, it was not plug and play, you had to be somewhat skilled at the
> switch you were configuring and good with a 110/66 block and a punch too.
> Many times you needed a couple of people to bust out the toner to make sure
> you had the right pair.
>
>
>> 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.
>>
>
> If I were you and I never will be, but I would try to adapt to the paradigm
> that VoIP is not behind the times, it is just trying to accommodate other
> technologies that are behind the times, such as modems and FAXs.
>
>
>>
>>  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.
>
>
> My mother can use a computer very well (and she was "trained" on a
> typewriter) and my niece is amazing with technnology for her age.
>
> Things get old and antiquated.  Morse Code, VCRs, reel to reels,
> switchboards, betamax, even OTA non digitial TV in a few months.  Everything
> changes, gets better, and people cling to the old ways.  I know several
> people that swear that their old records and record player sound better than
> a digital CD.
>
>
>> That's where their expectation is.  What you sell better sound and work,
>> in it's worst case, like the old TDM platforms did.
>
>
> And it does, at a MUCH cheaper price point, with many more features, if
> engineered and installed by someone who knows what they are doing.
> Flexibility is mind blowing.
>
>>
>>  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.
>
>
> I think you should try a few different vendors and solutions.  What you
> mention above is going to give you pretty much the best analog.  I am sure
> someone else can sell you something that does this with SIP.  My personal
> recommendation is Quintum, but I am sure others do just as well.
>
>
>>
>>
>>  What have people used in a small deployment, 2 to 4 FXS ports, that
>> REALLY performs like a traditional TDM delivered analog service?
>>
>
> Generally, you buy a Digium or Sangoma board with the right number of FXO
> ports and then in install Polycom (or whatever) SIP phones.
>
> Personally, with both Digium and Sangoma FXS ports, I get perfect
> operation.  Maybe try posting your configs instead of a book and biography.
>
> Have you explored your zap configs?  There could be a simple setting that
> makes everything wonderful.  Asterisk at home didn't really have much
> facility for that.
>
>
>>
>> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home
>> office Asterisk switch right now).
>>
>> Jim
>>
>>
> Thanks,
> Steve Totaro
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-biz/attachments/20080927/24bdbeff/attachment-0001.htm