similar to: Asterisk Language Status

? Hello-- Boy, it's been a long time since I posted to the user mailing list! Pardon me, I've posted this to dev also, but I thought the general users should also be aware of this. I'd like to announce a proposal to the Asterisk Community, that I introduced at Astridevcon last month. It is a new API for playing sound files (mainly speech). A pdf describing the Proposal in some

I'm not going to bore you with all the stuff I've done since November here. I put it, and some examples, in the file update1.txt in the git archive. To read it, do a git clone of https://github.com/WyoMurf/SaySentence.git I a nutshell, I've upgraded the SayScript grammar to handle expressions in the file names, upgraded the current en, fr, it, hu, and some others, to use the same

I see the following paragraph in the Asterisk trunk LICENSE file: "In addition, Asterisk implements two management/control protocols: the Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface (AGI). It is our belief that applications using these protocols to manage or control an Asterisk instance do not have to be licensed under the GPL or a compatible license, as we believe

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it be possible to write out the audio frames as they get recorded? Watson supports 16 bit signed little

Matt Riddell wrote: > >> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com >> <mailto:luca.pradovera at gmail.com>> wrote: >> >> I have been working on designs for two different projects, where both >> of them would need to use the IBM Watson streaming ASR service. >> >> Would it be possible to write out the audio frames

On Oct 22, 2014, at 11:47 AM, BJ Weschke <bweschke at btwtech.com> wrote: > On 10/22/14, 12:14 PM, Paul Albrecht wrote: >> On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: >> >>> Paul Albrecht wrote: >>>> Really? Shouldn?t something this major affecting the entire Asterisk >>>> community get discussed on the lists?

Hi, In Europe many Telcos implement power-save mode (See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information). Would you agree to have this feature added to the ones already discuused for next Asterisk release ? (See https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010) Regards -------------- next part -------------- An

I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017 I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K. When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo). After

Hi, I have to install CentOS 7 for a client, to act as cache & filtering proxy using Squid. I'd like to use this piece of specialized hardware : http://store.calexium.com/fr/systeme-pre-assemble/869-systeme-pre-assemble-rackmatrix-apu-amd-gx-412tc-quatre-coeurs-1-ghz.html There is no VGA or HDMI video output, just a serial port to connect to, and then three NICs and two USB ports.

crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.

I see that Asterisk distributes soundsets for English/English-AU/Spanish/French, and Russian. There is code for several other languages inside Asterisk; how does one obtain the other soundsets? Also, I noted that the source sound files don't seem to be publicly available for the sound sets that Asterisk distributes. I would assume that the source sounds are all 44khz (or more) cd quality

Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples:

http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're

This is long overdue (0.2.8 is out for about a week already), but anyway, here we go: ============ What''s this? ============ scRUBYt! is a very easy to learn and use, yet powerful Web scraping framework based on Hpricot and mechanize. It''s purpose is to free you from the drudgery of web page crawling, looking up HTML tags, attributes, XPaths, form names and other typical

In the past little while, we've seen a wave of attacks on asterisk, via the provisioning. It goes something like this: A. scan for IP phones on the internet, either via spotting something on port 5060, or via the port 80 web interface for the phone. Or, use web sites that scan the internet, and classify the machines, to make your work shorter. B. Once you get into the web GUI,

Hi, I''m encountering another highlighting issue. (about the first one "range search and highlighting", i received no response. I don''t even know if somebody tried to reproduce and/or if it''s normal behavior) about the new issue, an example will be easier for you to reproduce: I''m filling an index with random data, i try to match for "*1*"

Really? Shouldn?t something this major affecting the entire Asterisk community get discussed on the lists? Any idea what Leif is talking about when he says the community is in transition, moving from dial plan model to external control? Here?s a link to the notes posted on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2014

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Hi all, I''m wondering about my sanity here. There''s probably something simple I overlooked. First off, I have close to zero experience with Merb. 1. I configured use_orm :activerecord in config/dependencies.rb; 2. I have a file named app/models/measurement.rb which looks like this: class Measurement < Observation end 3. I have another file named

Rather hasty I think. I think whatever version 1.2.X winds up on should be the most stable release of Asterisk, period. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Olle E Johansson > Sent: Tuesday, May 29, 2007 5:19 AM >