twilio sip

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Create a sip environment in Windows

Create a sip environment in Windows Step 1 software preparation:Server Software ndosip server: http://www.brekeke.com/This software is written in Java. Therefore, install JDK before installation.Take sipphone as an example. Step 2: Install1. install ondosip server first. set a port during installation. this port is used to manage an http port of the server. therefore, you can define it yourself. set it to 8080. the installation type is for education

Multi-channel stress testing program for SIP proxy

This stress test is based on the basiccall Project Modification of resiprocate 1.5 and passes the resiprocate 1.5 proxy test. In theory, it should also be able to perform stress tests on other proxies. How to Use 1. Download resiprocate 1.5 2. Slave nodes/Resiprocate-1.5/resip/dum/test 3. Compile and run the basiccall Project Notes: 1. If you use the resiprocate proxy, pay attention to several parameters. Char * domains = "192.168.1.101 "; Char * interfaces = "

Handy Color picker Sip on Mac

There is always a lot of things, you just look at a glance has been fascinated. Measures have such an app to do is really the heart of the very, eye-pleasing, triggering the beauty of your inner impulse. Let's leave one, memo.It's so delicate.Simple operation, the upper left corner of the heart of the click can be anywhere to take color, the color point will zoom in.The color of the heart in the upper right corner, is the system of color board. Each color finish is automatically added to the Liu

Asterisk Configure PSTN analog card to make the SIP soft phone call out outside by PSTN fixed telephone

============================================== View hardware configuration # Dahdi_hardware==============================================View Dahdi Service ConfigurationMore/etc/dahdi/system.confShow the following content, obviously less my PSTN card configuration# Global DataLoadzone = usDefaultzone = usRebuilding the Dahdi service configuration#dahdi_genconfView Dahdi Service Configuration again# more/etc/dahdi/system.confShow# autogenerated by/usr/sbin/dahdi_genconf on Wed Aug 15 22:09:20 201

SIP Key-value Database (iii)--MONGODB distributed

SIP Key-value Database (iii)--MONGODB distributedThe random Read and write performance of a single-machine MongoDB is tested, and this section is about MongoDB distribution.MongoDB is distributed into two types, one is replication, and the other is sharding. We mainly look at sharding.Put a structure first:MongoDB auto-sharding configuration is very simple, in different machines to open Shard, config server, MONGOs process can be. (assuming that the c

Sip rfc 4538 authorization request through dialog

Background: Generally, when a UA receives a request that creates a dialog (invite/Subscribe/refer), it must decide whether to authorize the request, in some cases, UA determines whether to authenticate the request by determining whether the request is in a created dialog, For example, after invite creates a dialog, UA does not need to authenticate other requests (prack, Act, etc.) sent in this dialog, but the problem is that for refer, message, and SUBSCRIBE requests, A new dialog will be crea

The next day, learn about SIP (FreeSWITCH add recording function) (2)

Learn some of the FreeSWITCH core commands, and then learn more about FS in detail.To see if it was not previously suspected, two times when programming changes the configuration file, or Java injects some parameters into the configuration file, learn more about the following configuration file.This should be difficult, not clear.Ask Mr. Baidu.Learn a new knowledge of how FS adds recording functions to configureThe general telephone system can record voice calls in the system, and voice recordin

Differences between update and re-invite methods in SIP

In sip, both re-invite and update are used to modify the session parameter. The difference is that update does not affect the status of the dialog, and re-invite changes the status of the dialog. Therefore, update can be sent before the first invite is responded (that is, before 200ok of invite is received ). That is to say, update can be used to control early media. The re-invite can only be sent after the first invite cup responds (that is, after th

Conflict between MessageBox and sip on PDA Platform

InProgramWhen the MessageBox pop-up error message is applied, when you click the OK button, it is found that the keyboard icon on the mainmenu has disappeared. You must click it once to appear. Although it does not affect the use of normal functions, it may be confusing for PDA cainiao, and many methods to refresh the interface cannot be solved. Baidu cannot find the answer to this strange problem. You can directly query the problem on Google and find that foreigners also encounter this proble

Design and Implementation of the SIP protocol stack-Transaction Layer

A transaction is a request transaction sent by the customer (through the communication layer) to a server transaction, together with all the responses to the request of the server transaction, sent back to the client transaction. The transaction layer processes the re-sending of the application service layer, matches the response of the request, and the timeout of the application service layer. All tasks completed by a user agent client UAC are composed of a group of transactions. Generally, a

Comprehensive application of SIP and RTP 3

Based on the practices in the past few days, we have found an Optimal Configuration: 1. The SIP server uses trixbox. If you are familiar with Linux, we recommend that you use asterisk directly. 2 If the client is used directly, it is recommended that ekiga. By the way, how do I feel when using several clients: 1 Linphone: It seems to be well known. However, the latest version 3.1.2 crashes after being installed. I installed a general XP-SP3, compu

Python graphical interface Development programming: WxPython (SIP)

, has been completely attracted by WxPython , can not wait to practice, hereby record the individual learning process (Windows system):First, Wxpython environment installationThe most realistic and practical way to install Pip, and must be used in a manner that will synchronize the installation of dependent packages: pip install-u wxPythonsecond, Wxpython tastedAs a beginner, must not blindly directly into the subject, it is necessary to go through this stage, can be very good to help understand

SIP Key-value Database (i)--list NoSQL

versatile and most like relational database in a non-relational database. His support for the data structure is very loose, is similar to JSON Bson format, so can store more complex data types, he is mainly used to solve the massive data access efficiency problem. His storage seems to have a larger demand for disk space. The new version starts to support distributed. 4, Hypertable Hypertable and similar hbase are developed from Google's BigTable model, which is good for distributed support, bu

Thread sip blocking Queue __java in JAVA

Recently in the study of Java with the JDK and the contract, Java.util.concurrent, Discovery is very powerful, one of which is the work of many times the use of threading tool class Blockingqueue. During the actual development work and interview

sip:180 Ringing vs 183 Session Progress

If You know this phone is ringing (an ALERT q.931 message, for instance) you send a 180 ringing. If You receive a notification indicating this call is progressing, but you don't know for sure whether the user I s being alerted or not, your send a 183

Asterisk SIP Channel Driver Remote Crash Vulnerability

Release date: 2011-10-18Updated on: 2011-10-18 Affected Systems:Asterisk Open Source 10.xAsterisk Open Source 1.8.xDescription:--------------------------------------------------------------------------------Cve id: CVE-2011-4063 Asterisk is a free

Comprehensive application of 5-rtp packet Removal Process for SIP and RTP

The RTP receiving part is relatively simple (you do not need to consider jitterbuffer and so on). Start with here. In fact, there are three steps: 1. Create a UDP listener, such as 5200. 2. After receiving the RTP package, send it to the unpacking

Asterisk SIP Channel Driver DoS Vulnerability

Release date: 2012-04-23Updated on: 2012-04-24 Affected Systems:Asterisk 10.xAsterisk 1.xUnaffected system:Asterisk 10.3.1Asterisk 1.8.11.1Description:--------------------------------------------------------------------------------Bugtraq id: 53205

Automatic Registration of Sip

SipManager: setautoregisterandpolicy () --> SipService: openpolicysession () --> SipSessionGroupExt: opentoreceivecballs () opentoreceivenovel () --> AutoRegistrationProcess: start () --> first, perform the anti-registration (duration = 0)

sip:180 ringing vs 183 Session Progress

If You know this phone is ringing (an ALERT q.931 message, for instance) you send a ringing. If You receive a notification indicating then the call was progressing, but don't know for sure whether the user I s being alerted or not, you send a 183

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