similar to: [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

Hi all, I want to install a Medical Transcription Related S/W which is running in M$ Wallless Windows. I want it to be in my box. So does wine / Cedega / Cross Over supports Game port or com port emulation. S/w Name: Express Scribe Website : www.nch.com.au/scribe/ <http://www.nch.com.au/scribe/> Kernel : 2.6.8 Debian Sarge 3.1 Wine : 20050930 release This s/w uses game port for playing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA384 Hi, we are moving all our internal IT to an OSX server. Having used Dovecot on Linux for a couple of years now I want to transfer all IMAP folders so that users won't see a difference. Are there are any tips & tricks how to migrate Dovecot to Dovecot? I hope this is a simple thing to do. - -- Robert M. M?nch Mobil: +49 177 245 2802

On Thu, Aug 6, 2020 at 7:28 PM Dan Cropp <dan at amtelco.com> wrote: > Should the ARI DELETE /playback/{playbackId} be able to stop a playback > when a number is being played? > > > > Here is a test I am running. I am playing multiple portions (sounds and > numbers). > > > > curl -v -u asterisk:asterisk -X POST >

Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a number is being played? Here is a test I am running. I am playing multiple portions (sounds and numbers). curl -v -u asterisk:asterisk -X POST http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:789 If I attempt to

I have following dataset: > res [,1] [,2] [,3] [1,] 1946 4 1.27 [2,] 1946 5 1.27 [3,] 1946 6 1.27 [4,] 1946 7 1.27 [5,] 1946 8 1.52 [6,] 1946 9 1.52 [7,] 1946 10 1.52 [8,] 1946 11 1.52 [9,] 1946 12 1.62 [10,] 1947 1 1.62 [11,] 1947 2 1.62 [12,] 1947 3 1.62 [13,] 1947 4 1.87 [14,] 1947 5 1.87 [15,] 1947 6 1.87 Now I write following code

Ok, so here's the deal... I've been having a lot of trouble sleeping, so I've done some research and discovered this program called "Hemi-Sync". I decided to torrent it from the Pirate Bay to try it out before spending my money. What I torrented worked incredibly, so bought it, then torrented another Hemi-Sync program: Lucid Dreaming. It's a DVD-Audio program. What I

Hi all, I'm running ubuntu 8.04 and Wine 1.0. I am trying to get VEC foot pedals (http://www.nch.com.au/hardware/pedals.html - first one on list) to work with Express Scribe transcription from NCH (http://www.nch.com.au/software/linux.html). When I download the native linux version of the software, the USB pedals work fine but the application is unable to play wav files. Contacted NCH who

Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

Can anyone tell me how can I create echo test using ARI stasis application?

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek

Up till now, I have only used Asterisk versions 1.2, 10 and 11, on CentOS 4, 5 and 6, and have made extensive use of AMI and FastAGI connections to a multi-threaded backend written in C. For a new project, I am looking at trying Asterisk 16 with ARI, on CentOS 7. I was looking at the various ARI libraries available, particularly the ones for Python and Node.js in github. I noticed that the

Hello list, I am having trouble listening to GSM files created by Asterisk using a browser. I am noticing that some of my users succeed in listening to them and some others don't. I guess it is something of a codec problem that does not seem to be installed on all machines (though they are all WinXP). Anybody knows what one should do to listen to GSM files? I send files through the

Hi all, anyone never used PSGW as gateway beeween * and SkyPe? If yes, how does it works? How many session could I have on a single user ? Thanks all Giordano ThanksThis e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised

Le 2015-12-15 15:25, Joshua Colp a ?crit : > Sylvain Boily wrote: >> Hello, > > Just a note - there's an asterisk-app-dev mailing list[1] which is > better suited for these kind of posts. Ok > >> I did some tests because i'm interesting to transfer a non stasis bridge >> to a stasis bridge and i found a strange situation. > > You can't, you have

I am connecting to the ARI with subscribe all, so I can see channels being created. I now want to extract a variety of header variables (at the moment the from and to tag). I tried to read them from the ARI but Asterisk refuses since the channel is not in a stasis app. Is there a way to read these from either the ARI or AMI ? I'm trying not to modify the dialplan. Thanks Brian

Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael

Hi there! Is there any way I can turn on debug for ARI and sending the output to a separate log file? So far I have only been able to turn on ARI debugging in the console which results in the debug output being logged in /var/log/asterisk/messages I would love to have ARI debug log messages in /var/log/asterisk/debug or even better in it's own ari-debug file. With best regards Florian

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

Hello, I am experimenting with Asterisk 12.0.0 alpha1. I have a couple of SIP phones working. Good. I can retrieve data using curl to interact with the new Asterisk REST API (ARI). Good. Now I want to use the new ARI events API, which requires a WebSocket connection. I am using Node.js for the client, and have a stable connection to ARI events on the Asterisk 12 server. What I hope for is