dtmf ivr

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Analysis of the principle of DTMF

Transferred from: http://blog.csdn.net/wangwenwen/article/details/82649251. DTMF principleDTMF (double Tone mulitifrequency, dual tone multi-frequency) as a technology to achieve fast and reliable transmission of telephone numbers, it has strong anti-jamming ability and high transmission speed, so it can be widely used in telephone communication system. But most of the audio dialing is used as a telephone. In addition, it can be widely used in data co

FreeSWITCH IVR in Lua calls and executes Nodejs code

First, functional requirements:The corresponding script file is called through the FreeSWITCH IVR key, and Nodejs provides many modules, which can easily communicate with other systems or in any form, my application is to send HTTP POST requests via Nodejs;Because I'm not familiar with the way freeswitch directly transfers the execution of the Nodejs file, I execute a LUA script that executes a call to the Nodejs file in a LUA script and executes it w

Three modes of DTMF (Sipinfo,rfc2833,inband)

1, DTMF (dual-tone multi-frequency) Definition: By the high-frequency and low-frequency sound two sine wave synthesis represents the digital keys (0~9 * # A B C D).2. Methods for detecting DTMF data in sip: Sipinfo, RFC2833, Inband1) SipinfoFor out-of-band detection, DTMF data is transmitted through the SIP signaling channel. There is no uniform implementation st

Identifying DTMF signals using the Goertzel algorithm

value range between 32768 to 32767, if you get an already encoded media stream, such as g.711 encoding, then need to decode first), do the following calculation:When the above calculation is complete, we can get the energy value p of the frequency F reflected in these n samples:DTMF RecognitionThe above is the whole idea of Goertzel algorithm. If we want to use it for DTMF recognition, we still need to do some work.

VoIP DTMF notes

DTMF definition: Digital keys (0 ~ 9 * # a B C D ). There are usually three methods for detecting DTMF in VoIP: SIP info, inband, and out band (rfc2833). In addition, the latest RFC has been adopted for the requirements of DTMF In the 3GPP IMS specification.4733 replaces RFC 2833. 1. Sip info For out-of-band detection, DTMF

Simple implementation of pjsip DTMF, callback function: on_dtmf_digit

Pjsip source code is relatively powerful. Let's talk about what DTMF is? Dual-Tone Multi-frequency DTMF (Dual Tone Multi frequency), dual-Tone Multi-frequency, composed of high frequency groups and low frequency groups, each of which contains four frequencies. A high-frequency signal and a low-frequency signal are combined to form a combined signal, representing a number. The

DTMF-VAD Project Analysis

This item is the starting point for accurate identification of DTMF signals. After detecting this item, the DTMF decoder is triggered to decode the received data and wait until the corresponding dial key value is reached.1. analysis of input signal features: the signal is the circuit noise in the DTMF signal plus the channel. Now, preliminary analysis shows that

Use DTMF to enable asterisk to automatically call the extension

Background System Structure Extension 1026 and extension 1027 are used in the existing telephone system, and an asterisk is implemented under extension 1026 and extension 1027. A: In the 1027 system, ext. B: 1011 in the 1026 system, ext. 1012 is now called B through a. Because the language menu is implemented in the 1026 system, first, you can use the 91026 incoming call 1026 system (with an external dialing 9), and then enter 1012 according to the prompt. The existing vswitch does not sup

Principles and Implementation of VoIP DTMF inband

This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you! DTMF is called multi-tone dual-join, also called secondary dialing. There are three methods for VoIP to carry DTMF: inband, RFC 2833 (the latest RFC is 4733, which is referenced in IMS), and SIP info. The inband mode transfers the buffer generated by keys to the audio RTP stream, instead of definin

Set Chinese IVR in FreeSWITCH

FS does not load Chinese speech by default. The Chinese module needs to be compiled first in the SRC of FS. The unloading (do not restart FS) command is Make Mod_say_zh-install Then load the module on the FS console Load MoD _say_zh If you want FS to load the module every time it is started, The following examples 1. Store the Chinese voice packet in the/usr/local/freeswitch/sounds/en/us/callie directory. Name en. That is the/usr/local/freeswitch/sounds/en/us/callie/zh 2. Edit the file in/usr/

DTMF encoding transmitted by rfc2833 in Voice Transmission

DTMF encoding transmitted by rfc2833 in Voice Transmission 2007-03-23 11:13:48 The class is defined as follows: # If! Defined (afx_head_2833_h000093b5c358_6f19_475b_a49f_13bfcc9dfe1c00000000ded _)# DefineAfx_head_2833_h000093b5c358_6f19_475b_a49f_13bfcc9dfe1c00000000ded _ # If _ msc_ver> 1000# Pragma once# Endif // _ msc_ver> 1000 Class chead_2833{Public:Bool dispelhead (char *Cvalue );Int btodd (char * cvalue );Char * dtob (intNvalue );Che

Enghouse Interactive IVR Pro Remote Privilege Escalation Vulnerability

Release date:Updated on: Affected Systems:Enghouseinteractive Enghouse Interactive IVR ProDescription:--------------------------------------------------------------------------------Bugtraq id: 65000CVE (CAN) ID: CVE-2013-6838 Enghouse Interactive IVR Pro is the call center software. Enghouse Interactive IVR Pro 9.0.3 and other versions allow unauthenticated u

Design and Implementation of micro-kernel flow engine (IVR navigation) (I) -- Development Background

Development Background Our company is mainly engaged in the development of enterprise speech products, such as scheduling systems, command systems, teleconference systems, and call center systems. These systems share a common feature, that is, they involve calls, playing, sending and receiving buttons, and venue operations. Our business products are built on our softswitch system. The application servers of the softswitch system provide these services externally. Problems We found a

MySQL error './ivr/t_cdr ' is marked as crashed and should was repaired when doing LOCK TABLES

Label:This should be the case when running a business, MySQL database abnormal interruption caused the table exception, check the MySQL log error is as follows [Email protected] opt]# tail-n5/var/log/Mysqld.log16082713: in: -[Error]/usr/libexec/mysqld:table'./ivr/t_cdr' isMarked ascrashed and should be repaired16082713: in: +[Error]/usr/libexec/mysqld:table'./ivr/t_cdr' isMarked ascrashed and should be re

Design and Implementation of the micro-kernel Process Engine (IVR navigation)-General idea

Development Background Our company is mainly engaged in the development of enterprise speech products, such as scheduling systems, command systems, teleconference systems, and call center systems. These systems share a common feature, that is, they involve calls, playing, sending and receiving buttons, and venue operations. Our business products are built on our softswitch system. The application servers of the softswitch system provide these services externally. Problems We found a problem

RealPlayer malformed IVR object index Code Execution Vulnerability and repair

Affected Versions: Real Networks RealPlayer 11.0-11.1Vulnerability description: Cve id: The CVE-2010-2996RealPlayer is a popular multimedia player. When resolving RealMedia. IVR files containing malformed data headers, RealPlayer mistakenly trusts the indexes in the data structure to find Object List. If the attacker specifies an index outside the array boundary, the application will reference the object from the computed pointer and call it. Atta

ERROR 1203 (42000): User IVR already has more than ' max_user_connections ' active connections

Received a notice that the number of MySQL connection is not enough, let's see what the reason. Log in with Root to see that the current number of connections is far from the max_connections connection limit. Ask my colleagues to send me the wrong

Freeswitch configuration Wiki

Document directory C D G I L M M Cont. M Cont. P S X I have created a freeswitch kernel research and exchange group, 45211986. Welcome to join. In addition, we provide the SIP-based communication server and client solution. The following 95 pages are in this category, out of 95 total. C Channel Variables D Debugging freeswitch Deployment setup Dialplan Dialplan XML Download installation guide G Getting Started G

The VoIP-a reference guide to all things VoIP

and regualatory issues VoIP training: seminars, tutorials, on-line classes Protocols IP protocols: SIP, LTP, H.323, SCCP, MGCP, Megaco, IAX, stun, Enum, Trip, simple, RTP, pint, sctp, t.37, t.38, cops ITU protocols: SS7, ISUP ITU related standards: p.1010 OSP-Open Settlement Protocol Markup ages IVR presentation and dialog management: VoiceXML Call Control/conferencing/call routing: ccxml

Source Code address of the VoIP Open-Source Project (2) --- [all documents related to VoIP]

over satellite connections Nat and VoIP QoS-Quality of Service Packetcable Fax and VoIP VoIP codecs VOIP Bandwidth requirements How to debug and troubleshoot VoIP VoIP sites: VoIP sites on the Internet VoIP policy state and federal VoIP policy and regualatory issues VoIP training: seminars, tutorials, on-line classes Protocols IP protocols: SIP, LTP, H.323, SCCP, MGCP, Megaco, IAX, stun, Enum, Trip, simple, RTP, pint, sctp, t.37, t.38, cops ITU protocols: SS7, ISUP ITU re

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