[asterisk-biz] proper analog behavior using an ATA

Sun Sep 28 16:58:40 CDT 2008

Steve,

  Sorry if my email was packed with TMI.  It was not meant to annoy anyone.  I just wanted it clear I am not analog stupid and had a solid reference for ATA performance expectations.

> Just for future reference, you don't really compile Asterisk at home, it is just an image that you install.

  I said "Started with compiling my own and Asterisk at home." and should have said "started with compiling my own with raw editing of .conf files THEN trying Asterisk at home as a packaged solution".

Thanks for your comments and suggestions.

Jim

----- Original Message -----
From: "Steve Totaro" <stotaro at totarotechnologies.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz at lists.digium.com>
Sent: Saturday, September 27, 2008 2:40:17 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA

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 

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