dtmf ivr

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Upload work notes using PHP to write FTP files

$ Ftp_server quot ;*. *. *. * quot; $ ftp_user quot; lu quot; $ ftp_pass quot; loveyou quot; // setupaconnectionordie $ conn_idftp_connect ($ ftp_server) ordie ( quot; Couldntco $ Ftp_server = "*. *"; $ ftp_user = "lu"; $ ftp_pass = "love you ";// Set up a connection or die $ conn_id = ftp_connect ($ ftp_server) or die ("Couldnt connect to $ ftp_server ");$ Login_result = ftp_login ($ conn_id, $ ftp_user, $ ftp_pass );If ((! $ Conn_id) | (! $ Login_result) {echo "FTP connection has failed

PHP Development file download type

/x-freehand Fhc image/x-freehand Fif image/fif Fm application/x-maker Fpx image/x-fpx Fvi video/isivideo Gau chemical/x-gaussian-input Gca application/x-gca-compressed Gdb x-lml/x-gdb Gif image/gif Gps application/x-gps Gtar application/x-gtar Gz application/x-gzip Hdf application/x-hdf Hdm text/x-hdml Hdml text/x-hdml Hlp application/winhlp Hqx application/mac-binhex40 Htm text/html Html text/html Hts text/html Ice x-conference/x-cooltalk Ico application/octet-stream Ief image/ief Ifm image/gif

Use PHP to write FTP Files for Upload

Use PHP to write an FTP file and upload $ Ftp_server = "*.*.*.*";$ Ftp_user = "lu ";$ Ftp_pass = "love you "; // Set up a connection or die$ Conn_id = ftp_connect ($ ftp_server) or die ("Couldn't connect to $ ftp_server "); $ Login_result = ftp_login ($ conn_id, $ ftp_user, $ ftp_pass ); If ((! $ Conn_id) | (! $ Login_result )){Echo "FTP connection has failed! ";Echo "Attempted to connect to $ ftp_server for user $ ftp_user_name ";Exit;} Else {Echo "Connected to $ ftp_server, for user $ ftp_user

Win7 changing file types PHP file header-based document type validation class function

' = ' 0764743264647464 ',' db ' = ' d0cf11e0a1b11ae1 ',' PCX ' = ' 0a050101 ',' MP ' = ' 0CED ',' Doc ' = ' d0cf11e0a1b11ae1 ',' NRI ' = ' 0e4e65726f49534f ',' wks ' = ' FF00020004040554 ',' PF ' = ' 1100000053434341 ',' ntf ' = ' 4E49544630 ',' NSF ' = ' 4e45534d1a01 ',' Arc ' = ' 41724301 ',' Pak ' = ' 5041434B ',' ETH ' = ' 1a350100 ',' mkv ' = ' 1a45dfa393428288 ',' ws ' = ' 1d7d ',' GZ ' = ' 1f8b08 ',' Tar.z ' = ' 1fa0 ',' Ain ' = ' 2112 ',' lib ' = ' 213c617263683e0a ',' msi ' = ' d0cf11e

Introduction to PHP file uploading through FTP

to connect to $ ftp_server for user $ ftp_user_name ";Exit;} Else {Echo "Connected to $ ftp_server, for user $ ftp_user_name "; }// Try to login$ Filename = date ('ymd'). ". xml ";$ Source_file = "/usr/local/IVR/sendwireless/xml/data/". $ filename; // source addressEcho $ source_file;$ Destination_file = "/ITC/admin/logstat/ftplog/". $ filename; // target address$ Upload = ftp_put ($ conn_id, $ des

A VoIP operation support system has the general SQL injection and Arbitrary File Traversal Vulnerability (a large number of enterprises are affected)

A VoIP operation support system has the general SQL Injection Arbitrary File Traversal Vulnerability (a large number of enterprises are affected) Kunshi Network Technology Co., Ltd. develops a support system for small and medium-sized scale VoIP operation services. In addition to meeting the operating rate setting and package management requirements, in addition to basic functions such as account management, terminal management, Gateway management, data query, card management, number management

Basics of stm32

1. Development Tools. MDK and IVR are recommended for beginners. 2. Register development or database version development; register development first, and then combine the two; 3. Software Simulation or Development Board, first software simulation, and later combination of the two; Project Creation: 1. It consists of three parts: Start, user, Lib, output, and outlist; Outlist Directory: used to store generated intermediate files; Output directory

Goip connects with elastix through "Config by line"

Goip connects with elastix through "Config by line" by Grace Liu on May 17,201 3In elastix, Gateway For using config by line mode in goip, we will realize: 9 + 5 + number will go out via goip4's SIM card 1. 9 + 6 + number will go out via goip4's SIM card 2. Any callfrom goip will reach IVR of elastix. The detailed configuration is as follows: 1. Build a SIP Trunk as voptechgoip in elastix: The related content in peer details is: Context = from-PSTN

Mobile Internet SP Billing

Currently, a poor android application is charging fees. A large number of developers rely on built-in ad clicks and application promotion and download revenues are meager. Because Apple has such a platform, Apple developers have a lot of revenue. This section describes the SP billing method in Android, which provides you with an option to make money. SP billing is a billing method that directly deducts the fee from the user's mobile phone bill by sending SMS messages to the SP provider in the

China Telecom value-added business study note 13-value-added business management

Behavior Statistics and user cancellation. 3) User Registration/Cancellation Management Group users: the business system should be able to provide unified user interfaces to group users and support unified management such as registration and cancellation. Multiple registration/Cancellation methods are provided for individual users: Text message registration/logout IVR registration/Cancellation Web portal registration/logout Register or log out o

Asterisk card FAQ Summary

third-party modules. Dingming has released a strictly tested signaling module No. 7. Users who need it can download the trial from Dingming website.Http://www.dmlink.net/dowload2.aspfile name: asterisk 7 signaling ModuleQ: Does asterisk support h323?A: asterisk does not support h323. The asterisk-addon module includes several 323 modules. Dingming has launched a strictly tested h323 module. Users who need it can download the trial from Dingming website.Http://www.dmlink.net/dowload2.aspfile nam

Use the open platform of telecom capabilities to easily develop text message applications

China Telecom has recently opened its capability platform. Now this project is in beta1 state. Using the capability platform, developers can easily call the various capabilities provided by China Telecom through a unified API, including voice call (multi-party calls), SMS, IVR, and GpsOne positioning capabilities, and integrate these capabilities into their own special applications, through the MM platform of China Telecom, we promote, sell, and obtai

SIP applications (proxy, PBX ,...) Open-source

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

Lambda, an anonymous function of Python

Today, when I look at the example of Yate embedded python, I forgot what Lambda is, too faint! The tag is as follows. Example: def route(yate): def on_route(route): ... yate.onmsg("call.route", lambda m : m["called"] == "ivr").addCallback(on_route) Use Lambda in Python to create anonymous functions. Lambda comes from the lisp language. The Lambda format is as follows: Lambda arg1, arg2...: Lambda creates a function object, but d

Window Process opening from Windows Service

The IVR local business has a dba_bjldnw tool to run on the server. Once the business is disconnected, It is aborted. You need to write a daemon to monitor it, Restart once it is found to be disabled. To prevent the daemon from being disabled by users, I plan to run a Windows service in the background and list the current system process every five minutes. If If dba_bjldnw.exe is not found, restart. The problem is that the Windows service is not a wind

Asterisk + radius

must run the following command to compile this module .. /Configure -- prefix =/data/Asterisk -- Exec-Prefix =/data/Asterisk -- With-radius =/data/radiusclient-ng/3. run configure to set the initial conditions for compilation to specify the installation path. Specify the values of prefix and Exec-prefix. The second configure will instruct you to install asterisk to the/data/asterisk directory .. /Configure -- prefix =/data/Asterisk -- Exec-Prefix =/data/asterisk 4. run the make command to compi

In-depth introduction to asterisk (I): asterisk channel (channels)

situation when an incoming call arrives at asterisk: The arrival of a call is perceived by interfaces driven by a certain channel, such as sip socket; The channel driver creates a PBX channel for it and opens a PBX thread for the channel; Implement the dialing scheme; At this time, at least one of the following occurs: A. Asterisk responds to this call and plays media data or reads media data, such as IVR services. B. the dialing scheme p

Build and use a Linux development environment -- Use of common Linux commands

combination with-l, the file size is displayed in a user-friendly manner.Clear: clear the display on the terminal. You can also use the shortcut key Ctrl + l ("l" is a letter)Cd: Switch DirectoriesCd/mnt can enter the mnt (specified) DirectoryCd .. you can enter the directory on the previous LayerCd-enter the previous directoryCd ~ You can access the user's home directory (~ : Refers to the home directory. The home directory refers to the user directory in/home, not the pure/home directory)Pwd:

Build and use a Linux development environment -- Use of common Linux commands

/-ivr: copy the vim_configure folder to the code directory. Mv: Move a file or change the file name-V: displays the movement progress. Mv file1 file2 file1 renamed file2Mv vim_configure/code/vim_configure/move to the code directory Which: view the directory where the command is located Tar: compression and decompression command Gzip format (compressed package suffixed with .tar.gz ): Compression usage: tar cvzf compressed package name file 1 fil

VoIP bookmarks from Klaus Darilion

. License: LGPL; Homepage: sipfoundry.org. SIP Forum Test Framework (SFTF): A Framework to test SIP devices for common errors. License: GPL; Homepage: sipfoundry.org. Callflow: A powerful SIP call flow visualizer; Homepage: http://callflow.sourceforge.net /. SIP Scenario Generator: A powerful SIP call flow visualizer; Homepage: http://www.iptel.org /~ Sipsc /. SIPp: A powerful SIP performance testing tool named Sered by HP; Homepage: http://sipp.sourceforge.net /. SIP Applications (Proxy,

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