similar to: Report: NT4PDC to Samba3beta3 working

Heya all, So I've finally got a machine on this Samba 3 domain. Yey. Its an XP Pro box, so I grabbed the UsrMgr.exe and SrvMgr.exe programs from Microsoft, and hoped to at least use UsrMgr to see lists of users, maybe even as a full User manager, since I can't find a decent one that does all the samba3 stuff. Anyway, I'm on the domain, and I run the UsrMgr.exe program and I get an

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, First, as far as I can tell, this topic hasn't been mentioned in the mailing lists in awhile (atleast not as far as google can tell). Is there any client/program that can send "net start" requests to windows boxes. The ability to start a program as a service on a windows box remotely would obviously be very handy. I am aware of

Hi, I'm giving a look at [1] with this: cd /tmp git clone https://github.com/zaf/Asterisk-eSpeak.git cd Asterisk-eSpeak ln -s <path-to-asterisk-folder>/include/asterisk.h ln -s <path-to-asterisk-folder>/include/asterisk make I'm getting this: gcc -pipe -fPIC -Wall -Wextra -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o

Version 2.1 of app-espeak and app-flite modules just got released. Changes include voice selection for flite, support for 16kHz file playback in asterisk 1.6.x, use of libsamplerate for faster resampling, better error handling and cleaner code. You can get them here: Asterisk espeak module: http://zaf.github.com/Asterisk-eSpeak/ Asterisk flite module: http://zaf.github.com/Asterisk-Flite/

Hi, I recently discovered http://zaf.github.com/asterisk-mstts/ . In the page above, it is mentioned you have to subscribe to Microsoft Translator API on Azure Marketplace. In Azure Marketplace, I found something called "Microsoft Translator". This API is free within a 2 000 000 characters per onth limit. Is this the API needed for MS TTS ? If not, where and how can I find the good

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

I have written a module for asterisk that uses the eSpeak speech synthesizer (http://espeak.sourceforge.net/) to render text to speech. The source is available here: http://zaf.github.com/Asterisk-eSpeak/ It's similar to app_festival and app_flite. It's only tested against asterisk 1.6.1 on x86 Linux but it must be working for other 1.6 branches too. Comments, fixes and suggestion are

Hi Daniel On Mon, 2016-02-22 at 19:40 +0100, Daniel Heckl wrote: > try this http://zaf.github.io/asterisk-speech-recog/. > I have tested it myself, it works very well. I wanted to try it, but I obtain the following error message: "speech-recog.agi,en-US: API key is missing. Aborting. " :-( What am I missing? Any hint? Frank

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

Hi Marcelo, Thanks for replying, I do not know what this branch is. Could you please let me know. Also, I enabled google cloud speech API only from the console. Do I need more API enabled? On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhterres at gmail.com> wrote: > Did you already tried the cloud_api branch? > > Regards, > > Marcelo H. Terres <mhterres at

Hi, a set of AGI scripts that provide ASR and TTS for asterisk using the iSpeech API (http://www.ispeech.org/) are available on this page: http://zaf.github.io/asterisk-ispeech/ This is the first public release, updates will soon follow. Feel free to test and report. Regards, Lefteris Zafiris -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, FastAGI doesn't get (or deserve) much love these days but a lot of people are still widely using it. Here is a small project of mine, started trying to scratch my own itch, that some might find useful. https://github.com/zaf/agitator It is a reverse proxy for the FastAGI protocol with some interesting features. Most noticeably request based routing, HA/failover, load balancing and

Hi, I receive the following message after running "/etc/init.d/dovecot reload": Aug 06 02:00:11 dovecot: Warning: SIGHUP received - reloading configuration Aug 06 02:00:11 dovecot: Panic: file ioloop-iolist.c: line 25 (ioloop_iolist_add): assertion failed: (list->ios[idx] == NULL) I'm running dovecot version 1.2.13, compiled against uclibc on the Alpine Linux distro, version

Hi, The fact that apparently the last tigervnc update from upstream was missed triggered me to check for missing updates and packages in 6.3. Here are my results. Sorry for any false positives that might have crept in, but note that some of the 6_x updates actually are updates and not a parsing error. And perhaps an occasional false positive due to having to compare upstream SRPMS vs downstream

Hi, Reverting http://hg.dovecot.org/dovecot-1.2/raw-rev/2cdfcca171d8 solves this problem. Is there a preferred way to fix this issue? Duane Hughes -----Original Message----- From: dovecot-bounces+computers=za.jw.org at dovecot.org [mailto:dovecot-bounces+computers=za.jw.org at dovecot.org] On Behalf Of Computer Department (Watchtower South Africa Branch) Sent: 06 August 2010 09:59 To: dovecot

Hi Jonathan Thanks ! That would indeed be wonderful, at this point I really do not care whether I need to use Python or Lua or JS. I was following http://zaf.github.io/asterisk-speech-recog/ but hit a road end with (for the lack of sane word ) copulating Google's Key On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H <lardconcepts at gmail.com> wrote: > Yes! But I can only tell you if

Hello, I am very new in R so I am so sorry for this question. I have the Barro-Lee data set which contains 98 countries and I want to run the regressions only for the Latin America countries, so what do you recomend? How can I delete all the other countries or how can I select the countries of Lat. Am. thank you this is the list of countries SHCODE COUNTRY NAME WBCTRY

I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

Hi all, I'm trying to migrate NT PDC server to Samba PDC. and now i use samba 3.0.0beta2. I looks the migration steps from samba-howto-collection.pdf or in http://us2.samba.org/samba/ftp/cvs_current/docs/htmldocs/nt4migration.html, and it's not works well. The step is : ******** The approximate migration process is described below. " You will have an NT4 PDC that has the users,