[asterisk-biz] proper analog behavior using an ATA

[Jim Houser]

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