softphone | Python 3 compatible softphone with support for audio | TCP library

by DiscordPhone Python Version: Current License: MIT

X-Ray Key Features Code Snippets Community Discussions(10)Vulnerabilities Install Support

kandi X-RAY | softphone Summary

softphone is a Python library typically used in Networking, TCP applications. softphone has no bugs, it has build file available, it has a Permissive License and it has low support. However softphone has 4 vulnerabilities. You can download it from GitHub.

A python 3 compatible softphone built on the pjproject fork by malarinv, branch:py37. To use the streaming capability, use an Audio Callback class that suits your needs. See AudioCallbacks.py for examples.

Support

Quality

Security

License

Reuse

Support

softphone has a low active ecosystem.

It has 3 star(s) with 0 fork(s). There are 2 watchers for this library.

It had no major release in the last 6 months.

softphone 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 softphone is current.

Quality

softphone has 0 bugs and 0 code smells.

Security

softphone has 4 vulnerability issues reported (0 critical, 1 high, 3 medium, 0 low).

softphone code analysis shows 0 unresolved vulnerabilities.

There are 0 security hotspots that need review.

License

softphone is licensed under the MIT License. This license is Permissive.

Permissive licenses have the least restrictions, and you can use them in most projects.

Reuse

softphone releases are not available. You will need to build from source code and install.

Build file is available. You can build the component from source.

It has 377 lines of code, 36 functions and 8 files.

It has medium code complexity. Code complexity directly impacts maintainability of the code.

Top functions reviewed by kandi - BETA

kandi has reviewed softphone and discovered the below as its top functions. This is intended to give you an instant insight into softphone implemented functionality, and help decide if they suit your requirements.

Register a new SSH server
Wait for the signal to finish
Make a new call to an account
End the current call
Unregisters an account
Set null sound device
Destroy the audio stream
Create audio stream

Get all kandi verified functions for this library.

softphone Key Features

No Key Features are available at this moment for softphone.

softphone Examples and Code Snippets

No Code Snippets are available at this moment for softphone.

Community Discussions

Trending Discussions on softphone

Im trying something with IF in Batch but is not working as im expecting

Batch file parse phone numbers

While using Elastic SIP Trunking will Twilio still charge though the call was not received or answered?

Batch renaming files to a more meaningful name

Twilio SIP Domain not allowing inbound calls

Twilio | How can I modify call after X duration of dialed?

Code signing a framework fails with "Permission denied" error

Should I call pjsua_call_make_call (pj::Call::makeCall) in a background thread?

Display text as html in Razor page

How to login in RingCentral softphone in ipad ios in developer sandbox mode

QUESTION

Im trying something with IF in Batch but is not working as im expecting

Asked 2022-Feb-11 at 14:29

I need to do an script that looks for some text in a file and also look for a file itself in a %userprofile% path, and it works fine but when i tried to unify it so that instead of outputing 2 confirming messages only output 1, something is not going as expected, this is the code:

...

ANSWER

Answered 2022-Feb-11 at 14:29

The line in question has a '&' character which I assume you are trying to use as a boolean AND operator - 'IF' in CMD does not have an AND operator. The CMD command chain mechanism does and that's where the probably confusion arises. I typically nest IFs in cases like this:

Source https://stackoverflow.com/questions/71077611

QUESTION

Batch file parse phone numbers

Asked 2021-Jul-29 at 14:11

In order to use my softphone I have registered a custom url protocol, cccx

This is then associated with a .bat file that parses the phone and sends the command to the softphone.

The code I use is:

...

ANSWER

Answered 2021-Jul-29 at 14:11

The % character needs to be escaped by itself when attempting to use substring modification search replace syntax, which in turn requires that the substring modification be executed using delayed expansion.

Then all it takes to handle the cases presented is simple conditional test to work out what search replace actions are required:

Source https://stackoverflow.com/questions/68577134

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

Batch renaming files to a more meaningful name

Asked 2021-Jan-19 at 18:22

Soon I will begin receiving nightly zip file uploads that information will need to be renamed to make it more meaningful. Each night a zip file is uploaded and will be labeled similar to CallRecording_1-000.zip. Once extracted, it will contain the following set of files:

...

ANSWER

Answered 2021-Jan-19 at 18:22

@echo off
for %%z in ("ExtractFolder\AT1*.WAV") do (
  for /f tokens^=4^,8^,10^,12^ delims^=^" %%a in ('type "ExtractFolder\index.xml"^|find /i "%%~nxz"') do (
    for /f "tokens=1,2 delims=:" %%t in ("%%b") do (
      ECHO ren "%%z" "%%t-%%u_%%c_%%d%%~xz"
    )
  )
)

Source https://stackoverflow.com/questions/65779249

QUESTION

Twilio SIP Domain not allowing inbound calls

Asked 2021-Jan-13 at 06:31

I have purchased an US Phone Number on Twilio and associated it with a TwiML Bin.

The content of the TwiML Bin is supposed to redirect call to my SIP Domain:

...

ANSWER

Answered 2021-Jan-13 at 06:31

One thing you have wrong here is that when your TwiML bin dials your SIP endpoint, the webhook on the SIP domain will NOT be called. The SIP domain VoiceUrl will be called when you originate a call from the softphone that you have registered with the domain.

You've noticed that the console reports that your endpoint is not registered. Your softphone is responsible for registering itself and your username with the domain when it comes online.

I believe the issue you are having here is that you must specify the edge location in your softphone's domain configuration. Make sure you read the documentation on SIP registration.

Note that the bottom of that documentation has links to download a variety of step-by-step guides on configuring various softphones. I have actually tried to use Zoiper Desktop in the same way you are, but I found that it was not sufficiently compatible -- after successfully registering, it would continually re-register itself with the domain until it hit a rate limit, and then calls would fail. I've been very happy with Bria Solo as a reliable alternative.

Source https://stackoverflow.com/questions/65505646

QUESTION

Twilio | How can I modify call after X duration of dialed?

Asked 2020-Sep-14 at 12:09

Suppose a customer is calling and at the application side staff members(softphone) are busy with some other stuff. I want to play call later text when nobody will answer after X seconds.

I've used a timeout attribute of a Dial verb. But that directly ends the call.

Any help will make my day.

Thanks in advance.

...

ANSWER

Answered 2020-Sep-14 at 12:09

When the Dial timeout is completed, it will either hit the Dial action URL or if there is no action URL, the TwiML after the Dial (which isn't preferred since the TwiML after the Dial will play even if the dialed party answers).

Use the Dial action URL and the DialCallStatus to determine if the call was not answered during the timeout period, no-answer, and act accordingly.

Source https://stackoverflow.com/questions/63882801

QUESTION

Code signing a framework fails with "Permission denied" error

Asked 2020-Sep-08 at 08:30

I've recently added the Plantronics macOS SDK (Spokes3GSDK.framework) to my Electron desktop app.

When I attempt to sign the app, it fails with "Permission denied" error.

...

ANSWER

Answered 2020-Sep-08 at 08:30

The problem for me was that jenkins was copying the SDK files without changing the owner "root". So, a sudo was needed to make this command work.

I copied these files manually and put them in my repository (the owner of the file changed) and sudo was no longer needed.

Source https://stackoverflow.com/questions/63763588

QUESTION

Should I call pjsua_call_make_call (pj::Call::makeCall) in a background thread?

Asked 2020-Aug-07 at 13:42

I am developing a multi-account SIP softphone with custom media transport. I need to manage multiple calls simultaneously. I have read in "PJSUA2 Book" (3 PJSUA2-High Level API / General concepts / Asynchronous Operations):

(...) all operations that involve sending and receiving SIP messages are asynchronous, meaning that the function that invokes the operation will complete immediately (...) When this function [pj::Call::makeCall] returns successfully, it does not mean that the call has been established, but rather it means that the call has been initiated successfully. You will be given the report of the call progress and/or completion in the onCallState() callback method of Call class.

So I thought that I could call pj::Call::makeCall whenever/wherever I needed to initiate a call, the function would return nearly immediately and any progress would be reported in the callback methods. But today in my test environment I noticed a few seconds delay between the moment I called the function and it returned. Below there are detailed PJSIP logs that appear between makeCall is called and it returned. One can see almost 4s gap before "RTP socket reachable":

...

ANSWER

Answered 2020-Aug-07 at 13:42

After even more research I can reply to my own question. Maybe it could help someone...

I discovered makeCall triggered two DNS queries for localhost which failed after nearly 2 second each. That would make nearly 4 second gap.

The solution is to disable localhost IP address resolution completely by adding #define PJ_GETHOSTIP_DISABLE_LOCAL_RESOLUTION 1 into the pjlib/include/pj/config_site.h configuration file and recompile the PJSIP library.

The issue turned out to be already known and has been resolved in case of iOS: https://trac.pjsip.org/repos/ticket/1342

There was somebody else who experienced the same problem on Linux: https://www.spinics.net/lists/pjsip/msg20517.html

Source https://stackoverflow.com/questions/63299734

QUESTION

Display text as html in Razor page

Asked 2020-Jul-15 at 16:22

I've just started using Blazor and Razor pages.. it's a lot better than my old plain html pages I was building. But I'm having a problem converting my strings of html I build into actual HTML. Now, this works if I copy and paste my string and save that as an html page. It displays as html no problem. However, when I try to get my Razor page to display it, it displays as plain text. Even trying the trick:

...

ANSWER

Answered 2020-Jul-15 at 16:22

Wow ok.. so stumbled upon the answer:

Source https://stackoverflow.com/questions/62917124

QUESTION

How to login in RingCentral softphone in ipad ios in developer sandbox mode

Asked 2020-Jul-01 at 16:31

I have installed RingCentral Softphone in my ipad.

How to login in softphone in ipad ios in developer sandbox mode. I tried to check different options, but unable to find one

In Mac Os, I can switch to Sandbox via Cmd+F2 keys... but in iPad not finding solution

...

ANSWER

Answered 2020-Jul-01 at 16:31

I still think in iOS, it may not be possible for sandbox mode as it doesn't support.. RingCentral Phone mobile app does not support sandbox mode.

I can get this reference: https://forums.developers.ringcentral.com/questions/1289/is-it-possible-to-use-sandbox-account-with-ios-app.html

Source https://stackoverflow.com/questions/62679419

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

Vulnerabilities

No vulnerabilities reported

Install softphone

You can download it from GitHub.
You can use softphone like any standard Python library. You will need to make sure that you have a development environment consisting of a Python distribution including header files, a compiler, pip, and git installed. Make sure that your pip, setuptools, and wheel are up to date. When using pip it is generally recommended to install packages in a virtual environment to avoid changes to the system.

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 .

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