Touchstone
Posted: June 21, 2005 http://www.touchstone-inc.com/ VoIP testing and verification solutions for SIP and H.323 including call generation, analysis and QoS monitoring. Net Observer remote monitoring and diagnostic suite announced! Download your Free VoIP Analyzer HERE! Professional grade VoIP testing solutions starting at…
Read More »Asterisk cmd AGI
Posted: June 21, 2005 Synopsis: Executes an AGI compliant application Description: Executes an Asterisk Gateway Interface compliant program on a channel. AGI allows Asterisk to launch external programs written in any language to control a telephony channel, play audio, read DTMF digits,…
Read More »Asterisk cmd PHP
Posted: June 21, 2005 PHP() exten => 1,1,PHP(some.php|hello|world) Description Asterisk PHP allows you to control the dial-plan and write applications for Asterisk in PHP. This is faster and more flexible than phpAGI. res_php is located at http://eder.us/projects/asterisk_php/ See also Asterisk cmd AGI…
Read More »OrderlyCalls
Posted: June 20, 2005 OrderlyCalls is a multi-threaded telephony application server environment, in Java. You can run it stand-alone, inside your own application environment, or inside J2EE Servlet containers such as Tomcat. OrderlyCalls provides full support for the Asterisk FastAGI and Asterisk…
Read More »JAGIServer
Posted: June 20, 2005 JAGIServer JAGIServer has been replaced by OrderlyCalls. Full migration guide included. JAGIServer is produced by Orderly Software. JAGIServer is a 100% Pure Java application server for Asteriskusing the FastAGI protocol, which is a TCP/IP wrapper around theAsterisk Gateway…
Read More »Dial plan discussion
Posted: June 20, 2005 This page discusses the generic concept of dial plan, and compares it with numbering plan and routing plan. Numbering plan In its simplest form, a numbering plan allows for the addressing of elements in a voice network. It…
Read More »Asterisk FastAGI
Posted: June 20, 2005 FastAGI Implements the Asterisk Gateway Interface (AGI) over TCP sockets. This can help alleviate CPU load on your telephony server by relocating resource hungry scripts to another networked server. In order to instruct Asterisk to attempt a network…
Read More »Asterisk cmd MeetMe
Posted: June 20, 2005 MeetMe Synopsis MeetMe conference bridge Description MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe conference If the conference number is omitted, the user will be prompted to enter one. Page Contents Option details Option ‘s’ Option ‘q’ Option…
Read More »Mobicents
Posted: June 19, 2005 Mobicents is the First and Only Open Source JAIN SLEE 1.0 Certified product. Mobicents is a professional open source VoIP Middleware platform, which brings to telecom application developers what J2EE brings to Web and Enterprise application developers. In…
Read More »SFLphone
Posted: June 17, 2005 Overview SFLphone is a SIP/IAX2 compatible softphone for Linux. The SFLphone project’s goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed with a hundred-calls-a-day receptionist in mind….
Read More »exec
Posted: June 17, 2005 Usage: EXEC <application> <options> Executes with given <options>. Applications are the functions you use to create a dial plan in extensions.conf. Returns:failure: 200 result=-2success: 200 result=<ret> <ret> is whatever the application returns EXAMPLE: EXEC Dial Zap/g1/123456 Also, you…
Read More »SER module domain
Posted: June 17, 2005 Domain module implements checks that based on domain table determine if a host part of an URI is “local” or not. A “local” domain is one that the proxy is responsible for. Domain module operates in caching or…
Read More »get variable
Posted: June 17, 2005 Usage: GET VARIABLE <variablename> Does not work* with global variables. Does not work with some variables that are generated by modules.(* now works for global variables in 1.2.10 – see http://bugs.digium.com/view.php?id=7609 ) ‘Variable’ actually includes functions (but no…
Read More »Asterisk FWD Assistant
Posted: June 17, 2005 FWD Assistant A setup assistant to add a Free World Dialup account to the configuration of an Asterisk server running on MacOSX. Limitations The assistant is only aware of phones and extensions created with the New Extension Assistant….
Read More »SIP Info DTMF
Posted: June 17, 2005 The SIP INFO method can be used by SIP network elements to transmit digits out-of-band as telephone-events in a reliable manner independent of the media stream. The SIP INFO method contains a message body in addition to the…
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