pbx | Cloud PBX scripts for VoxImplant | TCP library

 by   voximplant JavaScript Version: Current License: No License

kandi X-RAY | pbx Summary

kandi X-RAY | pbx Summary

pbx is a JavaScript library typically used in Networking, TCP applications. pbx has no bugs, it has no vulnerabilities and it has low support. You can download it from GitHub.

#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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            kandi-support Support

              pbx has a low active ecosystem.
              It has 17 star(s) with 14 fork(s). There are 9 watchers for this library.
              OutlinedDot
              It had no major release in the last 6 months.
              pbx has no issues reported. There are 1 open pull requests and 0 closed requests.
              It has a neutral sentiment in the developer community.
              The latest version of pbx is current.

            kandi-Quality Quality

              pbx has 0 bugs and 0 code smells.

            kandi-Security Security

              pbx has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.
              pbx code analysis shows 0 unresolved vulnerabilities.
              There are 0 security hotspots that need review.

            kandi-License License

              pbx does not have a standard license declared.
              Check the repository for any license declaration and review the terms closely.
              OutlinedDot
              Without a license, all rights are reserved, and you cannot use the library in your applications.

            kandi-Reuse Reuse

              pbx releases are not available. You will need to build from source code and install.
              Installation instructions are available. Examples and code snippets are not available.

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            pbx Key Features

            No Key Features are available at this moment for pbx.

            pbx Examples and Code Snippets

            No Code Snippets are available at this moment for pbx.

            Community Discussions

            QUESTION

            How can I get a list of Azure AD Users in my PowerShell script within my Azure Function App?
            Asked 2021-Jun-08 at 21:11

            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 21:11

            Actually, 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

            Source https://stackoverflow.com/questions/67892345

            QUESTION

            Divs not vertically aligning inside flex container
            Asked 2021-May-27 at 10:30

            here is my code

            ...

            ANSWER

            Answered 2021-May-27 at 10:30

            1. Change your html structure

            Source https://stackoverflow.com/questions/67720085

            QUESTION

            sbt migration from 0.13.0 to 1.3.0
            Asked 2021-Apr-27 at 20:05

            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 20:05

            This 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

            Source https://stackoverflow.com/questions/67232850

            QUESTION

            Dropdown menu adding confusion
            Asked 2021-Apr-26 at 14:42

            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 14:42

            Figured it out, was not placing the new dropdown class I had created in the correct entry.

            Source https://stackoverflow.com/questions/67239213

            QUESTION

            PHP runs Perl script from command line successfully but I get no output displayed
            Asked 2021-Apr-07 at 18:44

            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 15:10

            I'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?

            Source https://stackoverflow.com/questions/66988154

            QUESTION

            While using Elastic SIP Trunking will Twilio still charge though the call was not received or answered?
            Asked 2021-Mar-30 at 22:14

            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 22:14

            For 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.

            Source https://stackoverflow.com/questions/66862986

            QUESTION

            Getting a 400 bad request error while submitting a flask form
            Asked 2021-Mar-27 at 05:22

            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 01:47

            Try 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.

            Source https://stackoverflow.com/questions/66826098

            QUESTION

            Asterisk - setup hangup after seconds dialplan otpion S(sec)
            Asked 2021-Mar-17 at 20:49

            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 20:49

            I think # mean "wait 1 second" so overal time is over 2 second. So "S" should work after D ends.

            Try following:

            Source https://stackoverflow.com/questions/66677016

            QUESTION

            With asyncio in Python 3 code, how can I (re)start/stop non-blocking websocket IO recurrently?
            Asked 2021-Mar-05 at 09:06

            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 09:06

            If 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-level asyncio package which exposes it publicly. You don't need to refer to the internal concurrent.futures._base module, which just happens to be where the class is defined by the implementation.

            Source https://stackoverflow.com/questions/66469586

            QUESTION

            Why the caller don't hear echo when I'm relpying the RTP packets I receied
            Asked 2021-Feb-08 at 12:13

            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 12:13

            In 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.

            Source https://stackoverflow.com/questions/66055522

            Community Discussions, Code Snippets contain sources that include Stack Exchange Network

            Vulnerabilities

            No vulnerabilities reported

            Install pbx

            After you successfully created and activated your Voximplant developer account you need to login to the Voximplant Control panel and complete these steps to have your PBX up and running:.
            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

            Support

            For any new features, suggestions and bugs create an issue on GitHub. If you have any questions check and ask questions on community page Stack Overflow .
            Find more information at:

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            CLONE
          • HTTPS

            https://github.com/voximplant/pbx.git

          • CLI

            gh repo clone voximplant/pbx

          • sshUrl

            git@github.com:voximplant/pbx.git

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