OpenVXI
From the home page: OpenVXI is a portable open source library that interprets the VoiceXML dialog markup language (www.voicexml.org). It closely follows the VoiceXML 2.1 specification, avoiding proprietary extensions of any kind. OpenVXI is only one component of…
Read More »Asterisk SIP trustrpid
trustrpid This defines whether or not Remote-Party-ID is trusted. It’s defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt Usage The following summarises information taken from Part 2 of this May 2010 blog post, based on testing v1.6: https://kb.smartvox.co.uk/asterisk/how-it-works/caller-id-in-sip-and-asterisk-part-1/ https://kb.smartvox.co.uk/asterisk/how-it-works/caller-id-in-sip-and-asterisk-part-2/ P-Asserted-Identity Asterisk does nothing…
Read More »iSoftel
iSoftel iRoute A Session Border Controller, with advanced route optimization engine, for Telecom Operators and Large Enterprises. iSoftel iRoute is vendor and protocol agnostic; adds business intelligence to the network infrastructure with advanced features such as Call Margin…
Read More »iCallGlobe – deleted
www.icallglobe.com is the World’s best Voice over IP (VoIP) Service Provider and one of the best leading suppliers of integrated communications and advanced and highly developed voice based services, to access the wholesale carriers globally. We are a…
Read More »LTP
Hotfoon’s Internally developed Lightweight Telephony Protocol LTP that enables enhanced telephony over data networks (like the Internet). The LTP based user agents are easily customized, skinned to suite various customer needs and deployment necessities. They are available for…
Read More »Asterisk cmd SetCDRUserField
Note: This function is depricated. It has been replaced by Set(CDR(userfield)=Value) in latest SVN. Synopsis SetCDRUserField(Value) Documentation SetCDRUserField sets the CDR, Call Data Record, user field to a value, that could be from a variable. CDR records are…
Read More »Asterisk redhat9 packages
Redhat 9 Package requirement The basic package requirement for Redhat 9 is pretty well documented but I wanted to know what would be the minimum package requirement. Here is a list of what I added to a minimum…
Read More »Asterisk readme iax
A copy of the README.IAX in the Asterisk distribution as of 09/22/2003: Inter-Asterisk eXchange Protocol INTRODUCTION This document is intended as an introduction to the Inter-Asterisk eXchange (or simply IAX ) protocol. It provides both a theoretical background…
Read More »Asterisk tips autoattendant
Autoattendant How can an Attendant switch on or off the AutoAttendant from her phone? Eg. 8am -> Attendent enters office -> switches OFF auto attendent. He/She takes in all the incoming calls and answers. 12pm -> out for…
Read More »Avaya
Avaya SIP Telephones: 46xx IP Telephone, 96xx IP Telephone, 16CC SIP CallCenter phone Communication Manager, SIP Enablement Services, Modular Messaging Avaya’s website 3rd Part SIP End Points with Avaya IP Office 500 Tel: +97143263939E-mail: info@epillars.comWebsite: www.epillars.com, ePillars http://www.iranvoipshop.com…
Read More »Asterisk cmd Read
Synopsis Read a variable in the form for DTMF tones as pressed by the caller Description As of Asterisk 1.0: Read(variable[|filename][|maxdigits][|option]) As of Asterisk 1.2: Read(variable[|filename][|maxdigits][|option][|attempts][|timeout]) As of Asterisk 1.4: Read(variable[,filename][,maxdigits][,option][,attempts][,timeout]) Reads a #-terminated string of digits a…
Read More »Asterisk Voicepulse Assistant
Voicepulse Assistant – coming soon A setup assistant to add a Voicepulse Connect 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…
Read More »Budgetone makering5
Budgetone custom ringtone Firmware 1.0.5.0 Actually, I think the format of the file has changed in version 1.0.50. I did some sniffing, and came up with an approach that worked for me. I’ve attached the modified version of…
Read More »Digital Acoustics
http://www.digitalacoustics.com/ Digital Acoustics Paging and Mass notification systems incorporate VoIP Speaker and VoIP Intercom endpoints that utilize existing network wiring. SIP Paging is routed via SIP Media Gateway product offering flexibility and integration with Asterisk systems. Endpoint models…
Read More »Guide to Getting Started with Asterisk on MacOSX
Asterisk source code distributions (release, svn trunk) compiles and runs on OS/X. If you are new to Asterisk and VoIP and you don’t have the time or the nerve to concern yourself with building software from source code…
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