SoftPhone | SoftPhone Ip Phone Asterisk Ed137 Pjsip | TCP library
kandi X-RAY | SoftPhone Summary
kandi X-RAY | SoftPhone Summary
Softphone is SIP-based virtual PBX software supports ed137 devices. Softphone has a Fast Fourier Transform (FFT) plotter sample on it, signal catch from rtp packets. There is a sample udp tcp communication inside the software.
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QUESTION
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 14:29The 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:
QUESTION
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 14:11The %
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:
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
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"
)
)
)
QUESTION
ANSWER
Answered 2021-Jan-13 at 06:31One 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.
QUESTION
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 12:09When 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.
QUESTION
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 08:30The 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.
QUESTION
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 13:42After 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
QUESTION
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 16:22Wow ok.. so stumbled upon the answer:
QUESTION
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 16:31I 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
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