DIgium Asterisk World turned out to be a good conference. Attendance was higher than I had anticpated, perhaps 150 people at the keynote and more than that aggregated amongst the different tracks and on the floor. Lot’s of interested parties, including enterprise buyers, vars, and a good showing by the business-oriented parts of the community, which was the goal of the event.
Our panel was well attended, and we had a good discussion about what “sophisticated services” mean from both a carrier and an application perspective. Moderator Bryan Johns did a good job and made some thoughtful comments. Co-panelist Chris Gatch talked about the benefits of using a next gen carrier and how asterisk as a pbx can take full advantage of such services.
Our slides are attached here, short and sweet. We overviewed different areas of sophisticated services possible on asterisk, such as conferencing, outbound campaigns, ACD/CTI, etc., then we focused on IVR and gave a short 4-slide overview of VXML, openVXI, and Voiceglue (including the voiceglue architecture diagram below — you can click thru to flickr for a bigger version). Several people asked some insightful questions, and in the post-panel discussions we had some requests for future project directions such as call control.
On last item of note is that we had a good talk with Craig Campbell of Cepstral, who donated a SDK so that we can integrate his TTS into Asterisk — which we plan to do as soon as we finish with our ASR integration. We do have a bit a of a roadmap that we’ve been putting together, and we’ll get a post out on that in the near future.
We’ve been asked to talk about using Asterisk to build scalable open-source IVRs at Pulver’s Fall VON and Digium Asterisk World. The conference is held here in Boston (our stomping grounds), and our panel is at 2pm on Tuesday, Oct 30.
I (Steve) will be on the panel along with a next gen service provider, and a very experienced Asterisk integrator. The title of the panel is Expanding to Sophisticated Services.
This is the inaugural year of the Asterisk World conference; which is the business-oriented counterpart to Astricon.
You can sign up for the full asterisk world conference, which gets you into all the keynotes and panels, or you can come to the expo only. As of today, you can still get a free expo pass if you go to the main VON site.
If you’re interested in Voiceglue, come talk to us at VON. Doug and I will be there all day Tuesday, and I’ll be back for Wednesday and part of Thursday as well.
Version 0.5 of Voiceglue is now available on the download page.
This version fixes a bug with validation of identifiers in VXML files.
If you’re getting errors like these:
Datatype error: Type:InvalidDatatypeValueException, Message:Value 'caller' does not match regular expression facet '([a-zA-Z]|[a-zA-Z$][a-zA-Z0-9_$]*[a-zA-Z0-9_])’
Failed to parse the URI’s content. Make sure that this document consists of valid VXML.
then you should definitely upgrade.
This version turns off Xerces XML validation, which reports valid
VXML documents (confirmed valid by XSV) as errors.
Voiceglue author Doug Campbell is planning to be at this year’s Astricon for the Thursday sessions and the first half of Friday. If you’d like to get in touch or meet with Doug, look for the tall geek muttering about VxML (just kidding, Doug generally doesn’t mutter), or drop us a line, contact details here.
Version 0.3 of Voiceglue is now available on the download page.
This version contains a one-line change to the voiceglue executable that handles newer versions of LibXML. These newer versions of LibXML give a different nodeName for text nodes than it used to. The symptom of this failing in the old version is that none of the TTS text gets heard.
It fixes a very nasty bug with cached grammars, that caused any cached grammar to terminate the script! Hard to get much done with that bug! Thanks to users on the mailing list for finding it.
Also, it removes install errors in cases where the special user accounts already exist.
The first release of Voiceglue is now available. We finished testing release 0.1 this afternoon, and it’s downloadable.
This first release supports DTMF input (no speech recognition yet), recording, and playback of TTS (via flite) and pre-recorded audio. We plan to integrate speech recognition soon using LumenVox’s engine and possibly Sphinx. Further details on features of this release are found here.
Lastly, to support the project and users, we’ve setup a mailing list.
Well, that last issue turned out to be a significant one. The base code was not properly processing SRGS grammars. We needed to fix the grammar parser to handle SRGS. We looked for an open source solution, couldn’t find one. So we had to write one from scratch. That work is now done. We simply need to do the final integration, some quick testing for obvious problems, and then package up the release. Alas, in the intervening time we had to shift resouces to a higher priority project. We are hoping to return to this last bit of effort in a couple weeks, in which case we should release by mid-May.
