voip ivr

I found a very good sip Forum: Huisi Communication Technology Forum Http://www.citiy.com A series of slides for SIP This handout is copyrighted by Zheng Yu. Allows copying, distributing, and customization under "GNU Free

, two-way callback, voice verification code, phone download and create a conference call and so on operations 3. Cloud communication initiates a request to the application server and makes corresponding actions based on the returned results. Commonly used for IVR related functions Let's start with the second kind of came David, because this operation is more capable of mastering the initiative. (‾??)?The first is a sub-account that can be use

Open source of SIP applications (proxy, PBX ,...) SIP Express Router (SER): Highspeed GNU sip proxy with a lot of features and a lot of ongoing development. homepage:Http://www.iptel.org/ser/. A really cool sip proxy-I like it! You can also take a look atDevelopment HomepageWith Web CVS. At the beginning you shoshould readAdmin GuideAndMailing lists Archive. Openser: A fork of the SIP Express Router, more development and new features. homepage:Http://openser.org/. Ser Media Server (SEMs): Me

instant messaging and real-time network capabilities into applications without having to build a server-side hardware environment. Melting cloud for developers of the different scenarios required to provide a range of products, technology solutions, including: Client IM components, client IM infrastructure, service-side rest API, client real-time network communication Base library. With these solutions, developers can build instant messaging products directly in their applications, and can crea

I recently bought a new office building and the telephone system of the new Office, including Softswitch, digital relay gateway, E1 and VoIP, which should be configured by me. Multi-function programmer :) The following are some records during freeswitch configuration. I will share them with you here. For more details, see the official FS and Chinese official website. 1. freeswitch SRC contains a Perl script. add_user adds users in batches. 2. view Use

Asterisk is a fully-software PBX system. It can run on Linux, BSD, Windows (simulated), and OS X. It provides all the features you want from the PBX, and is more than the PBX. Asterisk supports four types of VoIP protocols. By using relatively inexpensive hardware, it can interconnect with almost all standard-based telephone devices. Asterisk provides a directory-based voice mailbox Service (VOICEMAIL), teleconference, Interactive Voice Response (

With the popularization of VoIP technology, the security of VoIP voice communication has aroused more and more widespread concern in the industry. But where the security threats of VoIP originate, this should be the first step in the industry to systematically address VoIP security issues. In October of this year, VOI

Asterisk is an open-source VoIP system running on Linux. Basically, all traditional PBX functions are provided. Call features ADSI on-screen menu system Alarm Explorer Append message Authentication Automatic attendant Blacklists Blind Transfer Call detail records Call forward on busy Call forward on no answer Call forward variable Call monitoring Call parking Call queuing Call recording Call Retrieval

Asterisk is known as the Swiss army knife in the VoIP field, and radius is the AAA billing protocol. Also, let's see how asterisk + radius can be combined with each other... AsteriskIs an open-source software VoIP PBX system, which is a pure software implementation solution running in a Linux environment.AsteriskIt is a fully functional application that provides many telecom functions to turn your x86 machi

Due to the need of VoIP projects at work, I recently got in close contact with the asterisk engineering code. As for the great Position and Role of Asterisk in VoIP, I will not talk about it. Please refer to the previous article: astersik + radius simple strategy. The introduction of Asterisk materials in China, let alone the source code analysis of Asterisk, so I will share some of the information I usuall

different session timer settings. Connect two terminals with different media capabilities and different SDP messages by reporting in two different control sessions. To support different network types (V4, V6), and different transmission types, TCP/UDP/sctp/TLS C. Multi-Point call management In this scenario, multiple CPE (CPE is the abbreviation of "mermerpremiseequipment", literally referred to as "User front-end device") are connected to b2bua, and b2bua provides services for all CPE. D. i

value of is changed from G0 to 1.Then add a zap trunk and set the value of zap identifier (trunk name) to 2.Finally, modify route 9_outside settings in outbound routes: Set the trunk Sequence Value to zap/2, that is, we fixed the call transfer (outbound call) by line 2/fxo card) when we call line 1/fxo card, if call transfer is set, trixbox will be able to transfer the call to line 1 from line 2!At least till now, I still cannot transfer the phone number to line 1 from line 1. I personally thin

IOS background running WenyiI learned from the apple documentation that a general application can get a certain amount of time to run related tasks when entering the background, that is, it can run a short period of time in the background. Three other types can be run after, 1. Music 2. location 3. voip Article 2 Execution in the IOS background is the content described in this article. Most applications will be paused shortly after they enter the b

This article discusses: Basic knowledge of voice response applications Create a Voice response workflow Tips, keywords, and syntax Handling User Responses This article uses the following techniques: Speech Server 2007,.net Framework Embedded presence, instant messaging (IM), audio and video conferencing, and telephony are among the unified communications features provided by Microsoft®office Communications Server (OCS) 2007. Developers can build a set of OCS APIs to include these and othe

Many enterprise users usually deploy VOIP network phones as one of the most important considerations when deploying network applications. In terms of enterprise-level mature network application models, the trend of such VOIP phones becoming the mainstream communication applications of enterprises is becoming more and more obvious. In fact, according to a recent survey by Telappliant, 100 of the 41% companie

enhanced functions such as voice mailboxes, computer operators, operators, and Web portals. Support for the ringing group and line-finding group, which can easily achieve 800 of the enterprise; flexible and unique IVR voice response, voice mailbox and other personalized services, can solve problems that traditional voice systems cannot solve. The powerful function combination key enables more convenient and richer office applications. It provides per

What is Avaya exrience portal (AEP )? 1: a software platform that provides automated voice (or multimedia) user experience;2: A complete speech application system is developed based on the standard VXML (which defines how to use features such as speech recognition, speech synthesis, Internet access, database access, audio file playback, and DTMF input .) And ccxml (which defines a series of open standard call control APIs Based on XML;3: supports multimedia processing;4: fully web-based architec

"autofallthrough" is set to "yes ". this setting changes the previous rules so that the suspended call is immediately terminated in case of busy, blocking. If you are writing an extension for IVR, you must use the "waitexten" application. [General] configure several settings at the top of the Extentions. conf file. [Globals] then, in the [globals] section, you can define global variables/constants and their initial values. Contexts and extensions aft

Execution in the IOS background is the content described in this article. Most applications will be paused shortly after they enter the background status. In this state, the application does not execute any code and may delete it from the memory at any time. Applications provide specific services. Users can request the backend execution time to provide these services. Determine whether multithreading is supported Uidevice * Device = [uidevice currentdevice];Bool backgroundsupported = no;If ([dev

I. Git/gitlab create a new project above Get the project address: Http://git.mcc.com/mcc/voip-api-test.git two. Push local code to the remote repository Go to the local project path and push to the remote repository MINGSHEN@PC32 ~/git $ lssms/sms-client/voip-api-test/ MINGSHEN@PC32 ~/git $ cd voip-api-test/ MINGSHEN@PC32 ~/git/