Tag Archives: ATA

A new old phone and rotary dialing, revisited

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During the holidays I received and lightly restored another beautiful telephone from the 1940s, this Automatic Electric Monophone model 40:

ae40.jpgAutomatic Electric was an independent supplier of telephone equipment throughout the 20th century, eventually being purchased by GTE, and in direct competition with the Bell empire (or Western Electric, Bell’s equipment division) throughout its existence. The community-written Wikipedia article tells more of the story.

Some details about this Monophone:

  • Manufactured from 1939 to the mid-fifties, but I can’t find a stamp on the inside or outside to date it.
  • The coiled handset cable would typically indicate a later model, but the nicked-up brass screws inside the handset where the cable is connected suggest that the coiled cable was an upgrade or repair at some point in this phone’s lifetime. So it may be an old phone body with a newer cable, or it may be a later model, all-original.
  • The dial plate seems to be uncommon. Browsing many pictures of AE model 40s on the web, I haven’t seen any others with white-numbered, black dial plates. Also, it has no alphabet markings, only digits.
  • It’s all Bakelite, and heavy. The Northern Electric 302, with a thermoplastic shell, is almost a full pound lighter. External data: someone weighed their own collection of phones.

The Northern Electric 302 and the Automatic Electric Monophone 40 ready to duel:
ne320-ae40.jpgThe AE40 works fine and will be a new addition to the analog remix, but I decided to do one more search for VoIP analog telephony adapters (ATAs) that will accept rotary/pulse dialing. Recently, a commenter suggested UTStarcom adapters, but others report of an ATA that is still in service and accepts rotary dialing: the Grandstream HT-502/503. I have a 502 coming from an eBay auction and will report back on how it works out.

Analog remix part 1: a VoIP manual service exchange

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Recently, I decided to revisit the analog side of my home Asterisk VoIP system, specifically, the Northern Electric 302 that I keep hooked up and on display. It’s a conversation piece, for when folks spot the rotary-dial phone, they ask, “Does that thing work?” At that point I explain that I can answer calls on it, but because of the rotary dial (and because I haven’t fashioned or purchased a rotary-to-DTMF converter) I can’t place a call from it.

Until now. With a little tweaking of the analog adapter, a fresh rediscovery of VoiceXML through the Tropo application platform, and some straightforward CGI programming, I have a robotic operator ready to connect me to whatever number I tell her. No dialing necessary!
Speech recognition is not new technology, but I think that using it here is a clever way to get around the limitations of my old technology (the rotary dial) by using the computerized equivalent of even-older “technology” (the human operator). So this discussion on implementing a manual service exchange in VoIP is dedicated to Almon Strowger, inventor of the automatic telephone switch.
There are three components to this:
  • configuring the analog adapter as an automatic ringdown (hot-line) or warm-line
  • programming the speech-to-text “operator” service on Tropo
  • programming the call-back CGI on a web server that can connect to the Asterisk Manager and perform call-handling functions
listed in order of difficulty.
Part 1: the automatic ringdown
Most VoIP-analog adapters (and probably most IP phones for that matter) have the ability to act as a hot-line, which rings a set destination when the phone goes off-hook, or a warm-line, which does the same after a timeout period (for example, six seconds without any digits entered). If you’ve got your vintage analog gear on a dedicated ATA, go for the hot-line, because you’ll never actually be “dialing” anything. As for me, I’ve got my 302 bridged with a Panasonic cordless system, so I need to maintain the ability to dial. But after six seconds of no dialing, the ATA assumes I want the operator and connects the call appropriately.
The hot-line/warm-line is configured in the endpoint’s dial plan. Refer to your device’s manual. The common Cisco/Linksys/Sipura SPA adapters all seem to have the same dial plan syntax, documented here. Here are the key items from that document:

The following implements a Hot Line phone, which automatically calls 1 212 5551234.

The following provides a Warm Line to a local office operator (1000) after five seconds, unless a four-digit extension is dialed by the user.

In my case, my “operator” extension is going to be 1100, so part of my dial plan will contain | P6 <:1100> |, which tells the ATA to automatically connect the call to extension 1100 after six seconds of no input.
Now that the need to dial has been eliminated, we’ll need something useful at extension 1100, which will be the Tropo-powered speech synthesis and recognition system, discussed in a future post.