Author: VoIP Info
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…
Read More »VoIP International Legal Status
Posted: December 4, 2004 Mexico December 2004 In August the Mexican Telecom regulator Cofetel ruled that regulation on voice traffic applies without a regard of the technology been used. Therefore in order to provide VoIP services you need a public network telecommunications…
Read More »Asterisk tips wiki
Posted: December 3, 2004 Asterisk Audio wiki This is the product of me trying to get a better understanding of the dialplan. Use at you own risk. It will give you a wiki, that your users can add recordings to, delete them…
Read More »Telesthetic
Posted: December 2, 2004 Telesthetic’s Free Access numbers to VOIP networks All of these numbers are in the state of Michigan, and are provisioned off our Class 4/5 switch. If you have any problems with these numbers or would like your service…
Read More »Asterisk config dundi.conf
Posted: December 2, 2004 Configuration file for DUNDi lookups in Asterisk See also DUNDiLookup Application Asterisk config files DUNDi Enterprise Configuration SIP with no passwords – guide to set up DUNDi for SIP exchange Sample dundi.conf ; DUNDi configuration file ;…
Read More »Asterisk cmd SayPhonetic
Posted: November 30, 2004 Synopsis Syntax SayPhonetic(text) Description Sequentially plays each file /var/lib/asterisk/sounds/phonetic/character_p.gsm for each character in text. Essentially, this command spells the text outloud using the NATO Phonetic Alphebet. This also checks some other directories such as /var/lib/asterisk/sounds/digits/. It appears to…
Read More »Asterisk settings nufone
Posted: November 29, 2004 NuFone is out of business this page is of historical interest only. This is a very generic setup used to dial out from Nufone and to ring a SIP extension on incoming calls. The code is made entirely…
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