similar to: Integration of Google Speech API V2

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

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

I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190410/4c704231/attachment.html>

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

In other words. I there a way that both phones are ring with only one extension? On 06.02.19 15:05, basti wrote: > both phones are in the same net. > when the soft phone is shut down, on hardware phone only an led is > flashing to show an incoming call but no sound. > > both phones use the same extension. that is the reason why I use pjsip. > > On 06.02.19 14:59, Antony

Dear folks, I want to know whether the following things are right or wrong, in the first place. Then to know the right way to do the same, if it is wrong. I am having a table with the name "cd_details". using models, I am accessing data from this. my model is having the code like this. model filename : cd_detail.rb -------- class CdDetail < ActiveRecord::Base def self.find_data

Hi, I have a client connected to the ''net through 3 ISP''s. Have set up a Linux box to do routing and load sharing for the 3 connections. A fourth interface is connected to the LAN with private IP addresses. Am using iptables to SNAT traffic to the appropriate IP depending on the interface the packet gets routed onto. The setup looks something like this: Interface IP

On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote: > These two phones are not using the same extension, are they? If you shut down the softphone, does the hardware phone then ring? Antony. > On 2/6/2019 8:49 AM, basti wrote: > > both phones are registered. and the hardware phone can also make calls. > > but an incoming call is not displayed and also not hearing.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I''ve written a minimal sort of Perl module that dynamically creates and destroys traffic control rules for specific IPs. I''m currently using it for a user bandwidth control application at a client site. The module essentially gets Ethernet device(s), IP address and in/out speeds as input and dynamically creates classes, queues

Dear all, I have a doubt, How to fetch records from tables, which are not having ID field as its primary key. If possible, please tell me to way to extract data from those tables. -- with regds, Nahalingam N. Kanakavel. (http://www.nahalingam.bravehost.com/PlanetN/) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks

Hi, I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5 digits). ChanSpy is working fine for listening in to conversations initiated by these channels, and I can use '*' to randomly switch channels. However, is there any way in this scenario to be able to switch ChanSpy to a specific channel by typing in a ...# key sequence during a spy session?

Hello, I wish to write a label to the right-hand side of a plot (side=4) using mtext, with the text facing inwards - that is, rotated 180 degrees from the default orientation. How might I do this? (I've tried experimenting with las but no luck.) Thanks, Rashmi Rashmi Mathur Master's Candidate Fisheries Science and Management Research Group School of Resource and Environmental

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 Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

Hi. I am a little unclear about the ARM JIT support status. Is it working as of LLVM 3.1? If not, is it on the roadmap for LLVM 3.2? I am not currently interested in NEON support so if thats unimplemented, thats fine. thanks, Rahul

Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End

Hi, I am trying to understand and change the network buffering that is being used by Remus, the HA solution present in Xen. From what i understood from reading the code, Remus calls the postsuspend method of the BufferedNIC after it suspends the domain that sends TC_PLUG_CHECKPOINT message and start the buffering and then calls the commit method of BufferedNIC after it gets the acknowledgement