sip

Twilio: cannot rename subdomain null for SIP termination

由 杀马特。学长 韩版系。学妹 提交于 2019-12-10 23:32:30
问题 In relation to SIP registration, simply trying to add termination URI: Termination URI Configure a SIP Domain Name to uniquely identify your Termination SIP URI for this Trunk. This URI will be used by your communications infrastructure to direct SIP traffic towards Twilio. When you point your infrastructure toward this URI, Twilio uses a Geo DNS lookup to intelligently direct your traffic to our closest POP. Learn more about Termination Settings I can add <foo>.pstn.twilio.com fine, but get

SIP soft phone in java [closed]

由 微笑、不失礼 提交于 2019-12-10 20:51:41
问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 6 years ago . I am looking for the best open source softphone that is implemented in JAVA and comes in a jar file. Any suggestions? 回答1: No idea if it is best or not, but there is an applet http://www-x.antd.nist.gov/proj/iptel/ (look to JAIN-SIP-APPLET-PHONE). Also, our company has used that sip stack implementation for own

PeerUnavailableException using JAIN SIP API and the NIST implementation

由 有些话、适合烂在心里 提交于 2019-12-10 19:36:49
问题 I'm trying to build a simple SIP application using JAIN SIP 1.2 and the NIST implementation. I'm using JavaSE1.7 with Eclipse as my IDE. The problem: I am getting javax.sip.PeerUnavailableException when trying to construct a SipStack object. My main class: package net.bezeqint.sip.enp; public class ListenerMain { public static void main(String[] args) { try { System.out.println("Creating ExampleListener..."); ExampleListener listener = new ExampleListener(); } catch (Exception e) { e

SIP API media codecs

由 橙三吉。 提交于 2019-12-10 18:45:35
问题 I am trying to develop an android application with SIP API (provided in Android 2.3 or higher). I want to know which audio codecs and video codecs does this API supports ? does this API have any tools allowing to configure/choose codecs ? please help me. 回答1: I want to know which audio codecs and video codecs does this API supports? You can get supported codec list from docs itself, check AudioCodec. Those are: PCMU PCMA GSM_EFR GSM AMR does this API have any tools allowing to configure

Android SIP - recursive attempt to load library “/system/lib/librtp_jni.so”

由 狂风中的少年 提交于 2019-12-10 17:57:14
问题 I see above log message when initiating SIP call from Android app using Android SIP SDK. Here's the line which causes it: mSipManager.makeAudioCall(mSipProfileLocal, mSipProfilePeer, listener, 20); I don't think there's anything wrong (at least related to the above message) in that line. But anyway, after that method is called, I see recursive attempt to load library "/system/lib/librtp_jni.so" And SIP call never gets established. Also, I don't receive any error messages/exceptions - nothing.

video call using android

由 一曲冷凌霜 提交于 2019-12-10 15:42:43
问题 I want to develop a app which uses 3G for video calling where secondary camera will be use for video call. Is that possible to make video call? If it is possible please give me some reference or tutorial. 回答1: you should look for SIP protocol stack in android. There are various open source projects out there. look for SipDroid, IMSDroid is also a good example. And Sip Api is also available in Android after API level 9. 回答2: Another one worth looking at is the CSipSimple project. It is using

Asterisk : originate call doesn't set the CALLERID in the dialplan

由 独自空忆成欢 提交于 2019-12-10 14:54:13
问题 I am using Asterisk PBX to call a softphone, i use thise command : "originate SIP/100 extension 4004" , in the dialplan, I have to get the CALLERID variable, but in this case, it's always empty! P.S : if i call the extension (4004), from the softphone(100), the CALLERID is set, and I can get it with : ${CALLERID(num)} . How to get the caller id in the originate case? 回答1: When you're originating a call, you set the Caller ID yourself. There are two ways of doing this - either in the originate

Why is NDK generating shared library for x86 with text relocation even after setting -fPIC flag?

由 ∥☆過路亽．° 提交于 2019-12-10 14:40:16
问题 I'm using PJSIP/PJSUA2 with Android project. I've built libpjsua2.so shared library for armeabi, armeabi-v7 and x86 with -fPIC flag. The libs are also linked with OpenSSL 1.0.2h statically which were built with -fPIC flag. All libs loads properly on device with Android 4.4 to 6.0 except x86 in which it shows text relocation error. Build script builds two versions of the lib- android-9 (for API level upto 15 and no x86) android-16 (for API level 16+) Why on x86, text relocation is occurring

Qt5.5和PyQt5.5在UbuntuKylin15.04的快速安装脚本

由 。_饼干妹妹 提交于 2019-12-10 13:36:17
Qt5.5已经发布了，前两天PyQt也很快推出了一个5.5的对应版本。试验了一下（花了个周末啊！），真是越来越好用了。Qt5.5在Ubuntu15.04上有一些重要的改进，在虚拟机里运行的窗口覆盖问题也终于没有了。因为要装好几个软件，挺花功夫的。这里给出个脚本，可以快速安装。 首先安装Qt5.5。 #在线安装用这个. wget http://download.qt.io/official_releases/online_installers/qt-unified-linux-x64-online.run chmod +x qt-unified-linux-x64-online.run ./qt-unified-linux-x64-online.run #离线安装用这个. wget http://download.qt.io/official_releases/qt/5.5/5.5.0/qt-opensource-linux-x64-5.5.0-2.run chmod +x qt-opensource-linux-x64-5.5.0-2.run ./qt-opensource-linux-x64-5.5.0-2.run 然后，设置一下Qt的路径。 #Add Qt Path to /etc/profile. sudo gedit /etc/profile #add line:

UbuntuKylin15.04安装SM iObjects C++库

由 与世无争的帅哥 提交于 2019-12-10 13:30:27
GIScript2015版采用SIP对iObjects C++库进行封装，可以直接调用其功能。当然了，SIP的封装方法可以封装任意的C/C++库，你可以随意去扩展它，与各种各样的库联合进行使用，简单、灵活、自由而且强大无比。这里简单介绍iObjects C++库的在UbuntuKylin15.04上的安装，独一份哦！ 注意：目前的版本貌似设置与Ubuntu SDK有一些冲突（也许是我设置的问题），要使用直接QT的安装版即可。 一、下载QT和iObjects C++库 到qt.io去下载QT的社区版，目前测试的是QT5.4.2（5.5已发布，但还没测试） 。 SuperMap的iObjects C++库是一个功能强大的GIS库，以C++/Java/.NET等接口方式提供，可以支持64的Linux／Windows，是空间数据处理和分析、可视化的超强工具，可以在 http://www.supermap.com.cn 上联系获取。 二、安装iObjects C++库 首先安装一个许可文件的驱动程序，运行aksusbd-2.4.1-i386包里的dinst，使用: ./dinst 这是一个32位的程序，如果出错。去安装libc6-i386后一般就可以了。 iObjects C++库是免安装的绿色软件，解压到指定目录。 三、启动QT Creator 记得一定要用单独安装的QTCreator
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