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GIScript2016-Docker上的Jupyter Notebook部署

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【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> GIScript2016是支持Python3的地理空间数据处理和分析工具包。 GIScript2016支持Jupyter和Spark，可以运行在单机以及大规模集群之上。GIScript with Jupyter+Spark on Docker这一技术栈非常强大，通过conda包管理程序还可以安装大量的科学计算包，如NumPy、Scikit、Pandas以及OpenCV、NLTK、Tensorflow、Keras等机器学习软件包，实在是大数据处理、分析和深入研究的神器。 GIScript2016将C++系统库封装为Python的过程全部开源了出来，方便研究者使用C++编写自己的专业算法库，然后添加到这个框架中来，是编写高性能的Python扩展模块的极佳参考。下面我们在Docker中进行部署GIScript2016的方法，然后使用Jupyter Notebook进行基于浏览器的访问。一旦环境设置完毕，就可以在很多环境下部署Docker容器，使用非常方便。Jupyter Notebook是基于浏览器的编程“笔记本”，非常适合进行数据探索类的研究。 1.安装支持环境 1.1 安装Docker容器支持环境我用的Mac OS X，现在的Docker已经原生支持了，不用像以前要安装VirtualBox

Android NDK: load_library: cannot locate srand

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问题 I have an android project where I use native code to do stuff with SIP (using libosip2 and libeXosip2). My native code is compiled together with the libraries' sources into one module. The code compiles just fine and the generated library has all the symbols I expect it to have, but when I try to load the generated library I get the following error: E/eXosip.loadLibrary(9210): java.lang.UnsatisfiedLinkError: Cannot load library: reloc_library[1307]: 1941 cannot locate 'srand'... My

Android NDK: load_library: cannot locate srand

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Linphone for android is not working/missing libraries

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问题 I am trying to run linphone code which I get from git://git.linphone.org/linphone-android.git --recursive . After downloading it successfully, I tried to compile and run it as per the README file. I used Cygwin for Autotools, Autoconfig, Automake, aclocal, libtoolize and pkgconfig & Android ndk r8d. then I executed the prepare_sources.sh shell script in cygwin which downloaded some needed resuorces. After following all the steps, when I tried to run the code I get an UnsatisfiedLinkError

Sip Manager api support

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问题 I have gone through SIP Manager Documentation, it says - Not all Android-powered devices support VOIP calls using SIP. You should always call isVoipSupported() to verify that the device supports VOIP calling and isApiSupported() to verify that the device supports the SIP APIs. Your application must also request the INTERNET and USE_SIP permissions. I have Samsung galaxy young and ace mobiles, both are 2.3+, i have checked with the methods SipManager.isApiSupported() , SipManager

Display the name on incoming phone calls

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问题 I have a Java application (using Swing) that must display the details of the customer when a call is received. Is it possible to pass the phone number from a softphone (SIP) to my Java Swing application so that it can display the details? IS there any other way or program to do this? 回答1: For this you need to use Linphone SDK with your java application. You probably need to show the softphone from your java application or receive call in your application using SDK of linphone. 回答2: You have

Skype for Business - Response group SIP headers

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问题 We are running Skype For Business 2015 with EnterpriseVoice and want to allow calls coming through response group queues to be forwarded to mobile phones. We have developed a small app / service to facilitate that through replacement of SIP header names and values according to documentation in http://blog.greenl.ee/2011/12/30/modifying-sip-headers-managed-sip-application-api/ http://blog.greenl.ee/2013/12/16/response-groups-call-forwarding/. The replacement seems to work flawlessly, but Skype

Java SipServlet to build VOIP phone calls (between Computer and analog phone/mobile)

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问题 I'm interested in building VOIP that actually can be used to call to analog phone using SIP or H.323. But my question is that, is it even possible to build Computer to Phone & Phone to Computer VOIP phone calls with SIP or H.323? Else what is the most common way of achieving this task? I've successfully built an application that i can transfer voices between two computers by using socket, and my guess is that building SIP to communicate with analog phone is quite complicated (even though i

Setup TLS + ZRTP For VOIP Using Asterisk and CSipSimple

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问题 Im trying to setup voip exchange using asterisk ans CSipSimple as client, fol are the detials Server Side: Generate certificates for server and two clients Place the server cert in /etc/asterisk/keys/ sip.conf: [general] context=local allowguest=no alwaysauthreject=yes allow=gsm allow=ulaw allow=alaw directmedia=yes allowoverlap=no bindport=5061 tlsdontverifyserver=yes tlsenable=yes tlsbindaddr=192.168.0.119 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt

How to read Call-Info Header from Invite Message using sipml5

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问题 I use sipml5 with freeswitch and I need to detect when call should be answered automatically. The only part where I can get it from is SIP Invite message: recv=INVITE sip:username@IP:50598;transport=ws;intercom=true SIP/2.0 Via: SIP/2.0/WSS IP;branch=z9hG4bKd451.8dc49598935d4ebdf937de014cf1d922.0 From: "Device QuickCall"<sip:NUMBER@DOMAIN>;tag=68rtr6c12v9em To: <sip:michaltesar2@IP:50598;transport=ws> Contact: <sip:mod_sofia@IP:11000> Call-ID: dcd8fb4d69f0850840a743c152f4f7358a21-quickcall