telephony

Open source FSK decoder library? [closed]

你离开我真会死。 提交于 2019-12-22 04:13:32
问题 As it currently stands, this question is not a good fit for our Q&A format. We expect answers to be supported by facts, references, or expertise, but this question will likely solicit debate, arguments, polling, or extended discussion. If you feel that this question can be improved and possibly reopened, visit the help center for guidance. Closed 6 years ago . I'm looking for a library or tool to decode FSK in wav files, e.g. caller id. Currently using the tools bundled with vpb-driver for

How to make live voice phone call using Twilio instead of just playing an MP3 when call is answered?

风流意气都作罢 提交于 2019-12-21 06:08:17
问题 To call phone number from notebook through Twilio I created ASP.NET-MVC 5.2 application. I can call a number and answer the phone but I don't know how to achieve live voice(to be able to talk) connection instead of just playing music. I created an action method inside HomeController : public ActionResult Call(string to) { client = new TwilioRestClient(Settings.AccountSid, Settings.AuthToken); var result = client.InitiateOutboundCall(Settings.TwilioNumber, to, "http://twimlets.com/message

How to make live voice phone call using Twilio instead of just playing an MP3 when call is answered?

一笑奈何 提交于 2019-12-21 06:07:32
问题 To call phone number from notebook through Twilio I created ASP.NET-MVC 5.2 application. I can call a number and answer the phone but I don't know how to achieve live voice(to be able to talk) connection instead of just playing music. I created an action method inside HomeController : public ActionResult Call(string to) { client = new TwilioRestClient(Settings.AccountSid, Settings.AuthToken); var result = client.InitiateOutboundCall(Settings.TwilioNumber, to, "http://twimlets.com/message

How to know if a call has been established?

夙愿已清 提交于 2019-12-21 05:22:34
问题 Well, I did a program to make a call to any number of my contact list and it works like a charm. The problem is that I would like to know if the connection has been estabilished (the other person has picked up the phone), and I have not any idea about how to do it. Is there something like an interrupt that can alert me when the connection has been estabilished? 回答1: listen to the broadcast ACTION_PHONE_STATE_CHANGED, for more information check this out here 来源: https://stackoverflow.com

How to detect a call Drop in android

南笙酒味 提交于 2019-12-20 01:49:07
问题 I'm writing an app that runs on background during a telephone conversation and logs the coordinates to a file after the conversation has end,I know Android telephony API can detect a call manual disconnect by user*(correct me,if I'm wrong)*, But what I want is to know whether the service disconnection have caused due to call drop,is there a way or an API I can use to achieve this, All what I need is to programmatically differentiate a disconnected call and a dropped call. Please help. 回答1:

IBM Worklight - Unable to get network signal strength in Android

强颜欢笑 提交于 2019-12-19 11:36:11
问题 Below is my Java implementation. The idea is to display myUtilString in an edit text in order to see the signal strength in a Hybrid android app while in a WL.NativePage, but I get nothing. I have tested the app on a real device, changing to 2G, 3G and LTE but I don't receive anything. I have implemented a function called onSignalStrengthsChanged , but I cant figure out what is wrong. In LogCat there is nothing of interest; maybe onSignalStrengthsChanged never is fired. package com

RaspberryPI: Making SIP outbound calls using linphonec or an alternative SIP soft phone

主宰稳场 提交于 2019-12-19 04:15:18
问题 In my project I want my Raspberry PI to dial an outbound PSTN number. I have a SIP account with an assigned regular phone number and I can make the mentioned outbound calls on OSX using Telephone.app: Now I tried to achieve the same on my raspberry PI using linphonec First I register my SIP Provider with linphone with seems to succeed: $ sudo apt-get install linphone $ linphonec linphonec> register sip:4100000004@free4.voipgateway.org free4.voipgateway.org <PASSWORD> Registration on sip:free4

Originate a call with Asterisk - without the originating extension ringing

徘徊边缘 提交于 2019-12-19 03:40:06
问题 I have a completely standard installation of Trixbox with 2 SIP extensions set up on it. Both extensions are Snom 370 SIP phones. I can originate a call from one extension to the other using the following CLI command: originate sip/101 extension 102 This causes the phone on 101 to ring, then when that phone is picked up it dials 102. What I would like is for the phone on 101 to automatically call 102, without 101 waiting to be picked up. Is this something that can be done, or do the SIP

how to access com.android.internal.telephony.CallManager?

北城余情 提交于 2019-12-18 13:29:37
问题 I am trying to access CallManager class object from com.android.internal.telephony package. Here is my code: ClassLoader classLoader = TestActivity.class.getClassLoader(); final ClassLoader classLoader = this.getClass().getClassLoader(); try { final Class<?> classCallManager = classLoader.loadClass("com.android.internal.telephony.CallManager"); Log.i("TestActivity", classCallManager); } catch (final ClassNotFoundException e) { Log.e("TestActivity", e); } Unfortunately, this is throwing a

how to access com.android.internal.telephony.CallManager?

蓝咒 提交于 2019-12-18 13:28:11
问题 I am trying to access CallManager class object from com.android.internal.telephony package. Here is my code: ClassLoader classLoader = TestActivity.class.getClassLoader(); final ClassLoader classLoader = this.getClass().getClassLoader(); try { final Class<?> classCallManager = classLoader.loadClass("com.android.internal.telephony.CallManager"); Log.i("TestActivity", classCallManager); } catch (final ClassNotFoundException e) { Log.e("TestActivity", e); } Unfortunately, this is throwing a