pbx | Cloud PBX scripts for VoxImplant | TCP library
kandi X-RAY | pbx Summary
kandi X-RAY | pbx Summary
#VoxEngine scenarios for Cloud PBX functionality. This project lets developers implement Cloud PBX using the Voximplant platform. This README file describes how to use the provided VoxEngine scenarios and Voximplant Application Rules to process calls according to standard PBX scenarios: IVR for incoming call, forwarding call to operators after greeting playback, calls between PBX extensions, calls from PBX extensions to phone numbers, call recording, etc. The only thing you need to start building your Cloud PBX is a Voximplant developer account, you can get it for free at
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QUESTION
Some context: I have a PowerShell script that gets information about users and their licenses on Azure, and then saves that information to CSV file. It works locally. My goal is to have this script automatically run on Azure (I'm trying to do it in an Azure Function App) once a month, and then have the created CSV file be emailed to a specified email. However all I want to figure out right now is how to get the list of users so that the script can at least just run without errors.
I have very little experience with PowerShell and Azure Function Apps, so I'm stuck on a few errors I'm getting. I have spent the last few days troubleshooting to no luck.
Here is the beginning of the script that I can run from my local PowerShell:
...ANSWER
Answered 2021-Jun-08 at 21:11Actually, AzureAD needs to be imported a bit differently - it's been a problem for a while per this github issue. This seemed to work for most people:
- Setting the application to run as x64 bit: Function App> Configuration > General Settings > Platform > 64 Bit
- Setting the app to run on Powershell 7 instead of 6 on this thread
- Use:
Import-Module AzureAD -UseWindowsPowerShell
QUESTION
here is my code
...ANSWER
Answered 2021-May-27 at 10:301. Change your html structure
QUESTION
I am getting error while migrating sbt from 0.13.0 to 1.3.0. I am currently facing issue in error: not found: value scriptClasspath.
My build.sbt file after the migration.
...ANSWER
Answered 2021-Apr-27 at 20:05This is the first error that looks like a lead:
error: value +: is not a member of sbt.io.PathFinder scriptedClasspath ~= { cp => "modules/" +: "customer-modules/" +: cp }
It says that scriptedClasspath
is a PathFinder
and you're trying to add elements to it as if it was a Seq[String]
.
Read the docs on how to work with Path Finders and see the Scaladoc for the PathFinder
type.
Most probably you will need to adjust it to something like
QUESTION
I am banging my head against a wall trying to add a dropdown menu to an existing menu bar and can't seem to get it to line up right without screwing up the existing menu, can anyone please just add a single dropdown to my menu so I can see what I have been missing? I believe I am not getting the position: absolute/relative correct or maybe it is the paddings?
This is my code:
...ANSWER
Answered 2021-Apr-26 at 14:42Figured it out, was not placing the new dropdown class I had created in the correct entry.
QUESTION
I have a problem displaying the results of a Perl script that I am calling from my PHP webpage. The Perl script constantly monitors a socket and will display the output of this when run from the command line and also saves the output to a file. I know the Perl script is being called and running successfully as the text file is being updated but I do not get the output on the webpage as I was hoping for.
I have tried using the system(), exec(), passthru() and they all allow the Perl script to run but still with no output on the webpage so I am obviously missing something. Am I using the correct functions? Is there a parameter that I need to add to one of the above to push the output back to the webpage that calls the Perl script?
One example of what I have tried from the PHP manual pages:
...ANSWER
Answered 2021-Apr-07 at 15:10I'm no PHP expert, but I guess that exec
waits for the external program to finish executing before populating the $output
and $return
variables and returning.
You say that your sql.pl
program "constantly monitors a socket". That sounds like it doesn't actually exit until the user closes it (perhaps with a Ctrl-C or a Ctrl-Z). So, presumably, your PHP code sits there waiting for your Perl program to exit - but it never does.
So I think there are a few approaches I'd investigate.
- Does
sql.pl
have a command-line option that tells it to run once and then quit? - Does PHP have a way to send a Ctrl-C or Ctrl-Z to
sql.pl
a second or so after you've started it? - Does PHP have a way to deal with external programs that never end? Can you open a pipe to the external process and read output from it a line at a time?
QUESTION
We've built a softphone system with Twilio Rest API and quickstart.js. But we are facing now a vital issue that Twilio is charging us for just an established call though it is not received! And we got the answer here LINK . So we searched for the solution and found out "Elastic SIP Trunking" that need some VoIP system or IP-PBX configurations.
But we are still confused and the question arises here, "While using Elastic SIP Trunking will Twilio still charge as the same though the call was not received or answered?" .
...ANSWER
Answered 2021-Mar-30 at 22:14For Programmable Voice, there are usually 2 call legs. With the Twilio client, you have the leg into Twilio (where you tell Twilio via TwiML what to do next) and another leg to the PSTN (your TwiML told Twilio to dial the PSTN). Twilio bridges the two call legs together. Each call leg has a per minute cost associated with it.
In your example, the call leg from the Twilio client to Twilio is always successful/answered. The leg to the PSTN may or may not be successful. If the PSTN leg is busy or not answered, there is no charge for that call leg, only the Twilio client to Twilio leg incurs a cost.
How Much am I Charged for Call Forwarding with Twilio?
For the most part, Elastic SIP Trunking does not have call legs. You do need a call server/PBX to use Elastic SIP Trunks. The call server/PBX handles all the application functionality vs. Twilio with Programmable Voice.
QUESTION
I'm currently writing an app in flask but I seem to stumble on this issue where I get a 400 bad request error everytime I post a form. Could anybody find the mistake I made? Appearently the error comes on the element with HTML name "oneaccess-customer-gateway".
Flask code:
...ANSWER
Answered 2021-Mar-27 at 01:47Try using request.form.get("name_of_field")
instead of just request.form["name_of_field"]
.
When you use request.form["name_of_field"]
, flask assumes that they key will always be there. you canavoid this by using the other one, or the try
and except
trick.
QUESTION
When you dial 876, asterisk pbx start a call, send some dtmf code but doesn't close the call after 2 seconds. The call need to be closed by the user.
...ANSWER
Answered 2021-Mar-17 at 20:49I think # mean "wait 1 second" so overal time is over 2 second. So "S" should work after D ends.
Try following:
QUESTION
In my live phone speech recognition project Python's asyncio
and websockets
modules are used basically to enable data exchange between client and server in asynchronous mode. The audio stream which to be recognized comes to the client from inside of a PBX channel (Asterisk PBX works for that) via a local wav
file that cumulates all data from answering call until hangup event. While conversation is going on, an async producer pushes chunks of call record (each of them no larger than 16 kB) to asyncio queue, so that a consumer coroutine can write data to buffer before sending to the recognition engine server (my pick is Vosk
instance with Kaldi
engine designed to connect using websocket interface). Once the buffer exceeds a specific capacity (for example it may be 288 kB), the data should be flushed to recognition by send
function and returned (as a transcript of the speech) by recv
. The real-time recognition does matter here, therefore I need to guarantee that socket operations like recv
will not halt both coroutines throughout websocket session (they should be able to keep queue-based data flow until the hangup event). Let's take a look at whole program, first of all there is a main
where an event loop gets instantiated as well as a couple of tasks:
ANSWER
Answered 2021-Mar-05 at 09:06If I understand the issue correctly, you probably want to replace await self.do_recognition()
with asyncio.create_task(self.do_recognition())
to make do_recognition
execute in the background. If you need to support Python 3.6 and earlier, you can use loop.create_task(...)
or asyncio.ensure_future(...)
, all of which in this case do the same thing.
When doing that you'll also need to extract the value of self._buffer
and pass it to do_recognition
as parameter, so that it can send the buffer contents independently of the new data that arrives.
Two notes unrelated to the question:
The code is accessing internal implementation attributes of queue, which should be avoided in production code because it can stop working at any point, even in a bugfix release of Python. Attributes that begin with
_
like_finished
and_unfinished_tasks
are not covered by backward compatibility guarantees and can be removed, renamed, or change meaning without notice.You can import
CancelledError
from the top-levelasyncio
package which exposes it publicly. You don't need to refer to the internalconcurrent.futures._base
module, which just happens to be where the class is defined by the implementation.
QUESTION
I'm trying to play audio for who is calling me all by using NodeJS.
He is using SIP and initiate a call. I was able to receive the call and record his audio.
Now I'm trying to use the RTP protocol to play back audio to him.
So what I tried is to save each RTP packet I got, and send it back as is. But I'm don't undestand why he don't hear nothing.
The INVITE command
...ANSWER
Answered 2021-Feb-08 at 12:13In other words: According to RTP protocol, is sending back the packet can work, or do I need to reconstruct them?
You need to construct a dedicated RTP header and add the received payload. SSRC and timestamp must be generated in a meaningful way or the receiver may discard those packets.
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Vulnerabilities
No vulnerabilities reported
Install pbx
Create 3 scenarios in Scenarios tab by copying and pasting the code provided ( pbx_in, pbx_local, pbx_out )
Create couple of users in Users tab, please remember that they will be registered on Voximplant as PBX extensions if you use SIP phones to make/receive calls
Customize scenarios according to your requirements and users you created
Create application in Applications tab and specify Rules for the application to launch scenarios according to the specified Rule's Pattern
Register your SIP phones using users credentials you specified (2nd step), full username will look like sip:USERNAME@APPNAME.ACCOUNTNAME.voximplant.com. They should act like PBX extensions, so you should be able to make calls between extensions and to phone numbers at the moment.
Depending on Rules you specified (4th step) you can also make a call to PBX from outside to test incoming call scenario. If you want to connect the existing phone number using SIP or allow receiving SIP calls, you should authorize IP address/network used for that at Settings->Security->SIP White List
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