It seems that there is a common thread with Digium and the direction they are headed. I hope things turn around.

[WM: So do we.]

I was also using Flite. It would hang up the line just after calling in. Everything had to be commented out for it to work.

I would like to see a feature like listening to the latest podcasts on an extension. Downloading the newest file and playing it.

[WM: Pay a quick visit to the PBX in a Flash Forums.]

That’s my opinion, for what it’s worth!

I’m really curious as to the validity of the claim that distributing an ISO of Asterisk’s GPL2 version would violate either the letter and/or spirit of the GPL2. The quote was qualified, which only lends me to suspect the validity even more. I’m no lawyer, but the PBXiaF project seems much more compliant than some others if you ask me.

[WM: PBX in a Flash complies with GPL2. The asserted claim was that we were including proprietary, non-GPL2 Cepstral object code in our distribution. That claim is false. What’s particularly disturbing is that someone who apparently thought otherwise would raise the matter by reportedly making such allegations to a third party rather than ever contacting us. So the open source folks scratch each other’s eyes out while the competition focuses on developing platforms upon which commercial organizations can build viable phone systems. Great business model, huh? Really makes you want to write another Asterisk application… for free!]

I was more encouraged after hearing this.

[WM: Hope you’re right. Funny that no one ever bothered to fix TTS before 1.4 was released. Only through the work of a couple of our folks and another fellow did any of it ever get working. I would hasten to add that Cepstral has been the big winner in all of this, and their contribution in furtherance of the Asterisk open source project has been exactly zero to date. Digium also has promised "a big huge response" to our article. Can’t wait. Stay tuned!]

[Update: I’ve now received the ‘big, huge response’ but we don’t publish private emails without permission. I’ve asked so we’ll see what we see. Suffice it to say, there is no mention of any support by Digium for TTS in Asterisk 1.6.]

I am building a very large TTS applicaton that will be used by 1/3 of the state of Louisiana where residents will receive automatic call backs for their inspection reports in real time. (It reads them the inspector’s notes and all.) Currently, I am building this with 1.4 so I am quite discouraged to think about what I might face in asterisk upgrades 🙁

– Run with the IAXy concept (my next few points focus on that). VOIP providers are making the FXO/PRI/ISDN stuff all a bit irrelevant. However, IAXY’s are a great solution for using low cost phones as portable extensions.

– Add DNS to the IAXy. Hell, open the source, and I’ll do it myself and submit the code back to you. IAXy’s are great for taking on the road, but not if you have a dynamic DNS.

– Add distinctive ring, and some other missing features to the IAXy. (Again, give me the code, and *I’ll* do it for you for free.)

– Make a two-line IAXy. Right now, I can buy a Linksys SPA that has two lines from Staples or FutureShop for half the price of the IAXy.

– Add an power-over-ethernet option for the IAXy. The power cube makes it a far less elegant solution for the corporate desktop.

– Develop DUNDI further; I don’t mind this being a commercial registry, if it works. I don’t know of anyone actually using it. Maybe add a PSTN gateway to it, become a provider (or partner with one). If Digium were a good VOIP provider (or closely partnered with one), a lot of Asterisk users would use them by default.

– Create (or license) a good SIP/IAX2 phone, with a good XML display. A IP phone that was designed around Asterisk’s features from the ground up (and was supported explicitly by asterisk), would truly rock. Make your AsteriskGUI or FreePBX or FOP-like switchboard work right on the display of phone.

– I say turf AsteriskGUI and throw your weight behind FreePBX. AsteriskGUI seems like a pale imitation of FreePBX. It’d make your bundled appliances less lame, too.

– Bundled appliances are fine, but most people who seek out an asterisk-like solution are not the type of people who want to pay a per-seat license. They’ll go to the phone company if that’s their style. Any asterisk system I set up for people costs $0 per additional extension (well, the cost of an IAXy or SIP phone). Per-user licensing is inconsistent with asterisk users’ needs. You’re not a traditional PBX supplier, stop trying to act like one.

– Make some network-based FXO’s (and multi FXS/FXO combinations) Linksys (by buying Sipura, I guess) has a nice selection of FXO/FXS devices in various combinations. They’re eating your lunch in this market. You should be able to dominate. I think the trend is away from PC cards, and into these network based FXS/FXO devices; don’t miss the boat. If Asterisk had some external IAX based network FXS/FXO devices (maybe even ones that accepted the existing FXO/FXS modules), I’d certainly buy them as the preferred device for Asterisk installations. Now, I’m seriously considering going primarily to Linksys gear.

– I love IAX as a protocol. So much less of a headache than SIP. Leverage that more. The only bit of hardware you’ve created that does anything IAX is the IAXy, and as I mentioned, you need to do more with that. I’ve read about one or two IP phones that support IAX, but the only information from these are on Chinese sites; I’m not about to order direct from China.

Well, that’s enough of a rant from now. Here’s hoping Digium can start innovating again…

Asterisk is getting a foothold in major installations. It either is, or has the ability to, totally shake up the foundations of telephony within that world. But deprecating commands rather than keeping them for backwards compatibility is amateurish and Mickey Mouse behaviour, and will alienate them from the corporate world pretty quickly.

I’m glad 1.4 is solid, and supported, since I doubt I’ll ever have the nerve to move to 1.6 for any real world customer.

Sadly, other folks doing asterisk-ish stuff, such as FreePBX, are *much* better at moving forward without breaking things. I’m not afraid to do a FreePBX module update on a client site. I’m terrified to do an asterisk code update on a client site, and that’s rather sad…

On another note, have other people thought about Digium and its future in general? I find that with the power of asterisk, and the prevalence of great SIP phones or ATA’s, and great VOIP providers, these days I’m needing digium hardware less and less. (Only when I want to interface to an FXO or FXS, and there are some good Network based alternatives for that these days.)

They seem to be progressing by reselling third parties bundled PBX’s, which isn’t as progressive as I’d expect from the likes of Digium. Maybe they’ve lost Mark’s original vision and drive. His ideas like DUNDI and such could be extended and commercialized quite nicely by Digium. Instead, all their "exciting new announcements" seem to just center around slogging third party bundled PBXs, which is a bit lame…

[WM: Couldn’t have said it better.]

However, the response to the support case is shocking. TTS may not be the most important thing in the world to some people, but Ward’s point is spot on; it used to work, so they should fix it.

Asterisk should have built-in tts functionality out of the box anyway. I wonder if we’ll get a direct response from Mark here?

[WM: Phone systems are designed to last much longer than 5 years. 10 to 15 is the average in most of the federal government, for example. In the case of Asterisk versions, there was less than a year between version 1.2 and version 1.4. They broke TTS in 1.4. And they seem to think that same timetable and same approach is a good idea (again) with 1.6.]