similar to: Asterisk 1.8 minimum modules/configuration

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

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:

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

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo

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

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:

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

Hi, For years I've been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

On Tue, 2016-02-23 at 17:06 +0000, Steve Howes wrote: > Google?... Yeah... searched "google voice recognition api asterisk", browsed though various results. Nothing helpful for a beginner, very confusing bla bla... Thanks anyway for your help. F.

I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To

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

Im looking for wifi sip phones that support auto provisioning and work flawlessly with atserisk. Can anyone suggest me some models?

Dear ALL, Any clues or tips for the following gdb messages. [root@localhost asterisk]# uname -a Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct 29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux localhost*CLI> show version Asterisk CVS-HEAD-09/22/04-11:19:09 built by root@localhost on a i686 running Linux [root@localhost asterisk]# gdb asterisk core.13089 GNU gdb Red Hat Linux

hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using "res_timing_dahdi" or I can use "res_timing_timerfd" to get some benefit if I upgrade to 1.8? thank a lot for

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

Hi all Sometimes (about every three months) some of my Asterisk 1.8 boxes will start running this message thousands of times in the CLI: [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12

So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a "make menuselect" in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) res_smdi gets built fine, dahdi is