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Asterisk Blog - Page 20

Asterisk AGI php

Using PHP with the Asterisk Gateway Interface (AGI) PHP Tips and Examples Script Location Asterisk expects to find scripts in the /var/lib/asterisk/agi-bin/ directory for AGI execution. Page Contents File Permissions The Shebang Read AGI Inputs Sending commands and…

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Asterisk cisco FXO

How to connect a Cisco Router with FXO module to Asterisk Cisco routers support FXO or FXS voice interface cards (aka VIC). Either SIP or H.323 can be used for call signalling. VICs are installed into low density…

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Asterisk config queues.conf

ACD (Automatic Call Distributor) distributes incoming calls in the order of arrival to the first available agent. The system answers each call immediately and, if necessary, holds it in a queue until it can be directed to the…

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Asterisk config cdr_mysql.conf

Configuration of database for billing with CDR records Sample cdr_mysql.conf This is a sample configuration that assumes the SQL server is local. ; ; Note – if the database server is hosted on the same machine as the…

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Asterisk cmd AgentCallbackLogin

Page Contents Synopsis Status Description Replacement solution without AEL More info Logging off the queue manually Automatic logon / logoff via extensions.conf (it is possible) Login / Logout using the Manager API Logging off the queue automatically Finding…

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Asterisk Jabber

jabber is a common name for XMPP-based instant-messaged and communication. Native jabber support in Asterisk The Jabber module in Asterisk (res_jabber) is available starting from the 1.4 series. Therefore, you can connect Asterisk as a client (or component)…

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Asterisk Manager Proxy

The Asterisk Manager interface is not really designed to interface with lots of different simultaneous clients. Some of the issues: Every client using the Manager interface sees data for all other clients (bandwidth/security/privacy) Asterisk Manager interface doesn’t cope…

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Asterisk standard extensions

Standard extensions a: Called when user presses ‘*’ during a voicemail greeting h: Hangup extension i: invalid extension o: Operator extension, used for operator exit by pressing zero in voicemail s: Start extension in context t: Timeout extension…

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Asterisk cmd Authenticate

Synopsis Authenticate a user with a dialed passcode Syntax 1.4- Authenticate(password[|options[|maxdigits]]) Authenticate(/passwdfile|[|options[|maxdigits]]) Authenticate(/db-keyfamily|d[options[|maxdigits]]) 1.6+ Authenticate(password[,options[,maxdigits]]) Authenticate(/passwdfile|[,options[,maxdigits]]) Authenticate(/db-keyfamily,d[options[,maxdigits]]) Options A string of options may be provided by concatenating any of the following letters: a – Set account code to…

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Asterisk sound files additional

These sounds were recorded by Allison Smith, who does PBX voicemail voices for hire and is the voice of Asterisk. She does a great job! You can order prompts yourself from Allison by visiting here. This library is…

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Asterisk cmd SetCallerPres

SetCallerPres Set CallerID Presentation This application performs the identical operation to CallingPres(), but accepts descriptive text names instead of numeric values. SetCallerPres(presentation): Set Caller*ID presentation on a call to a new value. Always returns 0. Valid presentations are:…

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Asterisk consultants Canada – Quebec

Aheeva Technology inc. Web site: https://www.aheeva.com/ E-mail:[email protected] Phone: (514) 223-2581 Products: Inbound, SoftPhone, Predictive dialer, Full digital recording Atelka Contact Center Solutions Web site: http://www.atelka.com/ E-mail:[email protected] Phone: (514) 448-4905 Products: Virtual Contact Center Communications Télésignal Inc Web site:…

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Asterisk Monitoring

Now that you’ve built Asterisk around your entire life it might be a good idea to make sure it stays up! Argus The current version of Argus has builtin support for monitoring SIP and IAX2, as well as…

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Asterisk sip insecure

Insecure (For asterisk 1.0.9 and above) port: ignore the port number where the request came from invite: don’t require authentication of incoming INVITEs port,invite: don’t require initial INVITE to authenticate and ignore the port where the request came…

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Asterisk Manager API Action Hangup

Asterisk Manager API Action: Hangup Action: HangupParameters: Channel SEND: ACTION: HangupChannel: SIP/x7065558529-99a0 RECEIVE: Event: UnlinkChannel1: SIP/127.180.254.109-44df88e8Channel2: SIP/x7065558529-99a0Uniqueid1: 1124989110.20473Uniqueid2: 1124989110.20474 Event: HangupChannel: SIP/x7065558529-99a0Uniqueid: 1124989110.20474Cause: 16 Event: HangupChannel: SIP/127.180.254.109-44df88e8Uniqueid: 1124989110.20473Cause: 16 Response: SuccessMessage: Channel Hungup Asterisk manager API

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