Tag: Voice over IP

Interesting thread on integrating Speech to Text with Asterisk and PBX in a Flash

by Admin

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.
An old microphone
Image via Wikipedia

I just wanted to call your attention to this thread on the PBX in a Flash forum:

Exploring Speech to Text

This thread explores the possibility of adding a simple speech to text demo, and also discusses the possibility of transcribing e-mails to text.  It uses Google’s speech recognition service, and it is free to use.  I doubt you would be free to use it in any commercial application, but for those that just like to tinker with new capabilities for your Asterisk server, you might find this interesting.

I haven’t personally tried it yet, but I will say that if you are using some FreePBX based distro other than PBX in a Flash, you may need to remove the calls to the Flite speech synthesizer (or install Flite support).  I hate Flite (I think the voice quality sucks harder than a black hole — okay, maybe I exaggerate a little, but Cepstral voice synthesis is much better than Flite — unfortunately Cepstral is not free) so I’m not going to tell you how to install it.  Flite’s only used in the demo in the first post so if you are trying to do something else (such as attempt voicemail transcription, as discussed in the thread) you probably don’t need it anyway.

What would be nice would be the ability to dial a code, record a short message, and then have a transcription e-mailed to the address you use for voicemail notifications.  THAT is something I’d actually use on occasion!

EDIT: After posting this, I tried a basic installation on a NON-PBX in a Flash system.  Besides removing the references to Flite, I found I had to do the following:

  1. Change ownership of /var/lib/asterisk/agi-bin/speech-recog.agi to  asterisk:asterisk
  2. Install the perl modules mentioned in the “use” statements in /var/lib/asterisk/agi-bin/speech-recog.agi
  3. Install flac (some users may also need to install sox, but I had installed that previously).

Asterisk hiding a useful feature in plain sight by giving it a "cute" name

by Admin

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.
easter eggs in the stage of painting
Easter Eggs (Image via Wikipedia)

Somewhere in FreePBX 2.7 or thereabouts, it became know that there was a feature of FreePBX Outbound Route dial patterns, were you could use a /CallerID extension. This (among other things) basically lets you limit the use of an Outbound Route to a particular extension or group of extensions.  It’s a very useful feature, but wasn’t widely announced or promoted at the time.  I finally figured out why.

Thing is, it’s NOT a FreePBX feature, it’s a feature of Asterisk.  Anywhere in an Asterisk dial plan where you have a line that starts with

exten => _somepattern,…

you can use the Caller ID modifier, like this:

exten => _somepattern/callerid,…

In which case the pattern won’t be matched unless the current Caller ID number (which on an internal call is the number of the calling extension) matches whatever you’ve replaced callerid with.  Callerid can itself be a number or a pattern.

The real kick in the head is that it appears this feature has been around for a LONG time.  It was definitely in Asterisk 1.4.  Yet virtually none of the documentation you see on Asterisk even mentions this feature.  It might as well have been an “Easter Egg” hidden in the software, for all anyone knew of it.  Well, I finally figured out why — the Asterisk folks hung a “cute” name on it, and it stuck.

They called it ex-girlfriend logic.  The idea is that you can use it to stop an ex-girlfriend from bothering a particular user on your system (at least in raw Asterisk, though I don’t think that’s directly supported in FreePBX).  Besides being a bit sexist, it’s also about the last terminology anyone would think to Google on if they were trying to find out about this feature.  So while people were writing third-party modules like Custom Contexts and Outbound Route Permissions in FreePBX, it now turns out that essentially the same basic functionality was there all along, but hardly anyone (at least in the FreePBX world) knew about it until around about the time of FreePBX 2.7 or so.  If you can find anything at all about this feature in “official” Asterisk documentation (that doesn’t include third-party sites!), you’re a better searcher than I.

Makes you wonder if there are any OTHER cool features in Asterisk that are hidden in plain sight, under unfortunate descriptive names that no one would ever think to use when searching for such a feature!

 

Why your brand new router may cause your VoIP to stop working

by Admin

 

Important
This is an edited version of a post that originally appeared on a blog called The Michigan Telephone Blog, which was written by a friend before he decided to stop blogging. It is reposted with his permission. Comments dated before the year 2013 were originally posted to his blog.
I quote directly from Voip-Info.org:

Many of today’s commercial routers implement SIP ALG (Application-level gateway), coming with this feature enabled by default. While ALG could help in solving NAT related problems, the fact is that many routers’ ALG implementations are wrong and break SIP.

The article goes on to explain why the implementation is broken, and how to disable it in several brands of routers.  Certain VoIP adapter manufacturers also recommend disabling this feature if you are having problems with SIP registration, not being able to receive a call or one-way audio.  But note that this issue can affect any type of SIP-based communications, regardless of hardware or software used.

EDIT (May, 2018): For information on another issue that may cause problems when you switch routers, see this DSLReports thread: SIP registration times.

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