Azatel
Posted: December 14, 2004 Azatel designs, manufactures and integrates SIP & MGCP related products to provide our partners with highly advanced Analogue Terminal Adapters (ATA) and cost effective SIP solutions. AZACALL200 Our flagship model, the Azacall200, has gained significant industry recognition from…
Read More »Asterisk config manager.conf
Posted: December 14, 2004 Configuration of the Asterisk manager API. Note that activating the manager API opens a TCP port. Make sure you know what you are doing. Maybe it’s enough to just activate this for the local loopback interface, 127.0.0.1. However,…
Read More »Asterisk Network Protocol
Posted: December 13, 2004 Asterisk Network Protocol This page is dedicated to what the future manager interface should be. A Asterisk Networked protocol could bring multiple potentials, here are a list of items. To start, here are brain storming Beable to share…
Read More »Hex Editor
Posted: December 13, 2004 An editor for editing binary files. There contents for the file are shown in Hexidecimal, although binary, decimal, octal, and ASCII display modes may also be available. Windows Hex Editors wxHexEditor http://wxhexeditor.sourceforge.net Opensource – GPL license Native Win32…
Read More »Configuring Cisco 12SP+/30 VIP phones with Asterisk
Posted: December 13, 2004 This page documents how you configure a Cisco 12SP+/30VIP phone with Asterisk. The Cisco 12SP+/30VIP IP Phones are EOL and they are not supported by Cisco anymore. Cisco states that they do not have any listings for this…
Read More »Asterisk config extensions.conf sorting
Posted: December 12, 2004 NOTE: All the explained in this page is not true (at least in Asterisk 1.4.X). For example: [example] exten => _918.,1,Dial(Zap/1/${EXTEN}) exten => _.,1,Dial(Zap/2/${EXTEN}) exten => h,1,Hangup The real sort order is: _918. _. h Major change from…
Read More »E1
Posted: December 12, 2004 E1 is a the European equivalent of a T1 but runs at 2.048Mbps and has 32 x 64Kbps channels DS0s where 30 is used for voice and two are used for syncronization and signaling. Alarms that can occur…
Read More »Asterisk bounty symmetic SIP invites
Posted: December 9, 2004 Bounty Paid: 12/8/2004 to Theo Zourzouvillys Originally documented in https://issues.asterisk.org/jira/browse/ASTERISK-2326. Currently if we have Asterisk SIP channel driver binding to all interfaces, and eth0 has many subnets attached to it (a primary 10.1.200.1, and then alias interfaces eth0:1…
Read More »CallXML
Posted: December 8, 2004 The CallXML platform is available exclusively from Voxeo. As a simple markup language with a short learning curve, it suits the needs of most telephony applications that use touchtone input, DTMF. CallXML allows for faster development of IVR…
Read More »CCXML
Posted: December 8, 2004 CCXML – or Call Control XML – is the W3C standard markup language for controlling how phone calls are placed, answered, transfered, conferenced, and more. CCXML works hand-in-hand with VoiceXML to provide a 100% standards and XML based…
Read More »SRGS
Posted: December 8, 2004 SRGS is a W3C markup language standard for speech recognition, used in the VoiceXML standard and other IVR / speech platforms. This document defines syntax for representing grammars for use in speech recognition so that developers can specify…
Read More »SSML
Posted: December 8, 2004 SSML is a W3C standard for speech synthesis markup, that is used in VoiceXML and other IVR / speech platforms. The Speech Synthesis Markup Language SSML Specification is designed to provide a rich, XML-based markup language for assisting…
Read More »SER nat support
Posted: December 6, 2004 From the SER Administrator’s guide NATs are worst things that ever happened to SIP. These devices are very popular because they help to conserve IP address space and save money charged for IP addresses. Unfortunately, they translate addresses…
Read More »Asterisk phone cisco iad2420
Posted: December 6, 2004 The Cisco IAD2420 is a family of devices that speak MGCP. Generally they are used to connect via a WAN interface to a call concentrator and deliver their voice channels in the form of analog lines or a…
Read More »AimSoft
Posted: December 4, 2004 Billing Software: AIMSOFT does not have a Billing Solution for Asterisk yet. But it is planned. Overview: AimSoft provides a comprehensive suite of Billing and Customer Care solutions & 24×7 support for Prepaid and Postpaid Telecom Service Providers…
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Phone System 