dtmf ivr

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MT6165 schematic data mt6165 reference Design mt6165 data sheet information

-DC converter-Temperature Measurement SubsystemHttps://bbs.usoftchina.com/thread-202816-1-1.htmlOther recommendedThe MT8880 is a single-chip DTMF transceiver with a call progress filter. It is made of Mitel Iso2-cmos process with low power consumption and high reliabilityMT8389 MediaTek MT8389 is based on the CORTEX-A7 architecture, using TSMC's 28nm LP process technology with a clock frequency of 1.2GHz and GPU graphics processor for PowerVR [email p

RFC Chinese Document

conversation: Channel ManagementRfc2813 delayed Internet conversation: Server ProtocolRfc2817 update to TLS in HTTP/1.1HTTP over rfc2818 TLSRfc2824 call process language framework and requirementsRfc2825 complex network: release of i18n, domain name, and other Internet protocolsRfc2829 LDAP authentication methodRfc2830 Lightweight Directory Access Protocol (V3): Transport Layer Security ExtensionRfc2833 is used for the RTP load format of DTMF digital

Telephone ordering system (1)

This series helps beginners learn about the development of a telephone ordering system based on a multi-layer C/S structure. Profit Simplified some specific requirements for beginners. Hope to be useful to new users and hope for deficiencies Please note. Main functions of the system: (1) Use the incoming call display box to display the customer information. The system automatically determines the correspondence between the current number and the customer, displays the customer information, and q

Basic settings of SIP Trunk in trixbox

Basic settings of SIP Trunk in trixbox Http://www.voclub.net/zone? Action-viewthread-tid-1065The basic settings of the SIP Trunk in trixbox are as follows: the extension can call the phone number through a SIP Trunk, and then ring the trunk number to the extension. Create a new SIP Trunk, provided that you have obtained a sip account that can be connected to an external line. Dial rules: X. Here I set the simplest X. Because X represents any number 0-9, and X represents any length, so this trun

G723.1 audio codec Solution

output is input to the formant postfilter. A gain scaling Unit maintains the energy at the input level of the formant postfilter. Applications:WiFi phones vowlanWireless GPRS edge systems.Personal CommunicationsWideband IP TelephonyAudio and video conferencingWideband IP TelephonyFeatures:Full and half duplex modes of operation.Passes ITU test vectors.Common compressed speech frame stream interface to support systems with multiple speech coders (g.729, g.728, g.726 et al ).Optimized for high pe

Asterisk 1.8 SIP protocol stack Analysis 2

; ast_rtp_instance_set_timeout Ast_rtp_instance_set_hold_timeout Ast_rtp_instance_set_prop Ast_rtp_instance_set_qos Do_setnat The above is a call process for some columns, which initializes the RTP information of this peer, including QoS, Nat mode, RTCP, and DTMF tasks. The check_peer_ OK function has done a lot of work .... Now that the verification is passed, the RTP information is initialized, And the handl_request_invite function is returned

VoIP in-depth: An Introduction to the SIP protocol, part 2, 3-4

, and bodies a message may include. A subscribe request wocould include the event name in the "event" header, so that a single user can send out multiple SUBSCRIBE requests, even to the same target. to check which event packages are supported, a new header called "allow-events" was defined. subscriptions have an expiration time, and a subscriber that wishes to keep the subscriopen must send another subscribe request within the Dialog before it expires. this dialog closes when the notifier sends

[China Telecom value-added business Study Notes] 10 value-added business based on business nodes provides technology

is connected to the board switch, which has both a signaling link and a connection; The switch analyzes the number based on the access code and routes the call to the corresponding service node. The business control function (SCF) is responsible for business control and execution, and the implementation of a business is completed by collaboration between the SCF and SSF; The service data function (SDF) provides business data for SCF; The service exchange function SSF and call control funct

Common GPRS commands

all software modifications to the moduleAutomatic att Automatic Testing SoftwareWhether the input character is visible in the response to ateReply to factory settings at F software and restore to factory settingsDisplay Settings at V display the current settings of some parametersAuthentication Information ATI displays authentication information for multiple modulesRegion Environment Description at + CCed users get region ParametersAutomatic receiving level display at + CCed extended to displ

Incallscreen. Java/updatescreen () Analysis

The function updates the screen when the call status changes. 1. First, determine whether a call menu can appear. 1. If the phone is idle, the menu cannot appear. 2. If there is a call 2.1 if there is no waiting call currently, the call menu is displayed. 2.2 otherwise, the menu cannot be displayed 2. If the meeting mode is used, update the meeting Panel. 3. Update the callcard according to the current call status, that is, the Avatar shown in the middle of the screen. 4. Whether to display the

New features in Windows Phone 8: Start another program from one application (File Association and Protocol Association)

;The following code must be added after B. Start third-party programs that support the MKV Protocol Windows. system. launcher. launchurchill iasync (New uri ("MKV: hellokitty ")); The associated URI retained by Windows phone8,Note: ":" Before the keyword Bing: [keyword] Open Bing and search by keyword callto: DTMF: http: [url] Open the specified urlhttps: [url] in the browser open the specified urlmaps: mailto: [email] Open the email interface and se

How XMPP works

solves the problem of how to establish a connection to an object protected by the firewall or NAT (network address translation. XEP-0177 jingle raw UDP Transport. The file describes how to establish a connection in the same network without a firewall. XEP-0180 jingle video content description format. Defines the video transmission process from one XMPP object to another. XEP-0181 jingle DTMF (Dual Tone Multi-frequency ). XEP-0183 jingle telepathy tra

The experience to config Cisco 2811 for VOIP

; # framing NO-CRC4 Set the mode to NO_CRC4 GW_1 All the time slots in one primary group GW_1 All time slots into the first group GW_1 GW_1 GW_1 Turn on isdn overlap-switching ing mode GW_1 Config the dial-peer GW_1 # configure terminal GW_1 100Voip Add a inbound route named 100 GW_1 28829... The number starts with 28829. The "." cocould be wildchar to match any charact

VoiceXML identity Element and Its Attributes

VoiceXML Element Assign values to variables. Play the audio file. No user interaction executableCodeBlock. Capture an event. Define a menu item. Clear one or more framework items. End a call. Used for the else in the Used for elseif in the Lists menu options. Capture an Exit a call.

Android-controls the volume and playback of multimedia applications

This article translated from: http://developer.android.com/training/managing-audio/volume-playback.html Good user experience is predictable. If your application needs to play multimedia, it is vital that you can use the hardware or software of your device to control the volume, such as Bluetooth headsets or microphones. Similarly, when appropriate, your application should provide media playback operations such as playing, stopping, suspending, skipping, and forward on the media stream. Identify

Principles of XMPP protocol

to establish a connection to an object protected by the firewall or NAT (network address translation. XEP-0177 jingle raw UDP Transport. The file describes how to establish a connection in the same network without a firewall. XEP-0180 jingle video content description format. Defines the video transmission process from one XMPP object to another. XEP-0181 jingle DTMF (Dual Tone Multi-frequency ). XEP-0183 jingle telepathy transport method. XMPP proto

List of meanings of default values in Android system

Def_pointer_speed Integer Pointer speed setting, range 7 to 7 Def_dtmf_tones_enabled bool Whether the dialer enables DTMF tones when dialing Def_sound_effects_enabled bool Whether to enable audio feedback such as touch screen unlock Def_stay_on_while_plugged_in bool Whether to stay awake when charging is plugged in Def_max_dhcp_retries I

Building Java EE network applications for different clients

located. The dedicated Server software module, called Gateway, translates the client's request into an HTTP request. For example, a WAP gateway is responsible for translating WSP requests into HTTP requests (and vice versa) as well as parsing and interpreting responses. Similarly, using a VoiceXML request, a gateway made up of a voice browser is responsible for identifying the language and DTMF input, converting it into a standard request format, an

Android system cancels automatic lock screen

, Settings.System.EMERGENCY_TONE, 0); Set Default CDMA Call Auto RetryLoadsetting (stmt, Settings.System.CALL_AUTO_RETRY, 0); Set Default CDMA DTMF typeLoadsetting (stmt, Settings.System.DTMF_TONE_TYPE_WHEN_DIALING, 0); Set Default Hearing aidLoadsetting (stmt, Settings.System.HEARING_AID, 0); Set Default TTY ModeLoadsetting (stmt, Settings.System.TTY_MODE, 0); Loadbooleansetting (stmt, Settings.System.AIRPLANE_MODE_ON,R.BOOL.DEF_AIRPLANE_MODE_ON); Lo

The principle of XMPP protocol

Format. The video transfer process from one XMPP entity to another is defined. XEP-0181 jingle DTMF (Dual Tone multi-frequency). XEP-0183 Jingle Telepathy transport method. XMPP Protocol Network Architecture XMPP is a typical C/s architecture, rather than using Peer-to-peer client-side architectures like most instant messaging software, this means that in most cases, when two clients communicate, their messages are passed through the server (with t

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