voip

We are facing an issue regarding callKit Framework by iOS. We have to implement following functionalities in app. One to One call (Working fine) . we can end and accept second call (Working fine) . we can hold and accept calls (max 2 calls). we can switch between calls. Hold/Unhold current call. Issue : The issues we are facing are : We are able to accept second call which have no audio when hold and accept. Switch call button from call kit is disabled. We have done following implementation for handling multiple calls : We are reporting new call by following method. - (void

Closed. This question is off-topic. It is not currently accepting answers. Learn more . Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 3 years ago . For those of you working in the voice space, what are you using as your IVR platform? I am using Microsoft Speech Server 2007. What are some equivalent packages? Is anyone using open source software for handling inbound or outbound calls? Note that I'm not just talking about speech recognition, which is one component of a comprehensive package. An IVR platform would consist of speech recognition,

I've been having issues with compiling pjsip for iOS 4.0. I am using the latest trunk version from SVN and keep getting a portaudio error. When using the piedmontwireless guide: http://www.piemontewireless.net/PJSip155_and_iPhoneSDK312 I get a missing separator error in my build.mak file, which would indicate a whitespace/tabbing error, but for the life of me I cannot find it. According to the pjsip mailing lists, you should be able to compile out of the box for iOS 4.0, but I get this error: ../src/pjmedia-audiodev/errno.c:23:26: error: portaudio.h: No such file or directory ../src/pjmedia

Im trying to run a VOIP call using built in SIP on android 3.1. I have physical tablet device (galaxy Tab 10.1). For testing purpose, I have created a project from SipDemo example - it works fine! (meaning my credentials are working and my device/network is fine). my Manifest.xml <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="modera.com.doorcontroller" android:versionCode="1" android:versionName="1.0"> <application android:icon="@drawable/logoeditedsmall" android:label="@string/app_name" android:debuggable="true"> <activity

Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI) , using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also instruct Asterisk what to do next to a call/conference. Are there parallels to AMI and AGI in FreeSWITCH? Michael Collins These are good questions. I just wanted to add a few things to what @dkwiebe said (which is correct, BTW). The AMI equivalent in FreeSWITCH is the event socket. (Technically it's not the "Event Socket Library" or ESL, which is an abstraction layer for writing programs that use the

Is it possible to use an Android phone as a simple GSM gateway? The phone would receive a VoiP call using (preferably) Android built-in SIP stack, initiate a GSM call, and bridge audio both ways. After one call is terminated, the other one ends, too. How could I approach the problem? My earlier attempts failed at bridging audio between connections. Is there a SDK supported way of doing this, that I missed? Or do I need to implement some sort of a workaround? There are two problems with what you are asking: How to get at the incoming audio stream of the cellular call. How to get at the outgoing

I am new to iphone development. I want to develop an VOIP application. Anybody give me some ideas to start.I have tried "siphon" and "telephone" open source projects but they did not run. Thanx for any suggestion. Brad Larson For developing iPhone VOIP applications, you will probably want to read the answers to the question " Open Source VoIP/SIP Objective-C Code ". As you seem to have already tried siphon, but were unable to get it to compile, you might be interested in the answers to the question " How to compile pjsip for iphone 3.0 " (pjsip is what is used for siphon). 来源： https:/

I'm trying to create a VoIP based IVR service that interacts with a web application. From what I understand, adhearson runs on top of asterisk. What else do I need to have on the server to satisfy the equation? I think I need a way for asterisk to connect to a voip account. I'd appreciate any help and/or phrases to google. If you want to build your IVR quickly and easily you'd be better off looking at something like tropo.com (tropo's parent, voxeo, own adhearsion) or twilio.com, they've done a lot of the grunt work for you and setting up Asterisk is not for the faint hearted. If you want