Author: VoIP Info
Planet Phones
Posted: December 19, 2003 PLANET Technology Corporation is a leading global provider of IP Telephony products and solutions for small-to-medium-sized businesses,enterprises, and SOHO markets. PLANET designs, deploys and delivers innovative IP solutions with converged date , voice, and video deployment over the…
Read More »Asterisk extensions
Posted: December 15, 2003 Asterisk extensions Extensions are where you call in Asterisk. It can be A user A Asterisk cmd Queue Voicemail Voicemail Administration A PSTN connection and much more. A extension can be defined in text or with numbers. Always…
Read More »Asterisk cmd SetAccount
Posted: December 14, 2003 Note: This function is deprecated. It has been replaced by Set(CDR(accountcode)=[account]) in latest SVN. Synopsis: Sets account code Description: SetAccount([account]) Set the channel account code for billing purposes. Return codes Always returns 0. The ACCOUNTCODE variable The accountcode…
Read More »Asterisk X11
Posted: December 13, 2003 Problem: I just got started with Asterisk and am having a problem with voice quality. When connecting via either a Grandstream IP phone or calling from the PSTN (via Digium 100XP FXO) the demo-congrats msg I get is…
Read More »Buena Vista Communications
Posted: December 7, 2003 We are experts in IP Telephony with a history in mission critical US military communications. We offer flat-rate domestic calling plans unbeatable international rates brandable calling card platforms advanced IP communications consulting services brandable billing services hosted billing…
Read More »Asterisk cmd WaitForRing
Posted: December 7, 2003 Synopsis Wait for Ring Application. “This application can help with phantom ringing caused by a noisy FXO line.”(quote taken from the dev mailinglist, stated in a conversation between Marius Stankevičius and Eric “ManxPower” Wieling on 03/26/2007) Description WaitForRing(timeout)…
Read More »OpenEnum
Posted: December 3, 2003 This is work in progress, brainstorming OpenEnum – global routing of VOIP calls based on numeric input Background reading Read ENUM, Asterisk config enum.conf, Asterisk cmd EnumLookup, SER module enum, Asterisk E164 Call Routing The problem: phones want…
Read More »SIP Express
Posted: November 29, 2003 The SIP Express Residential Gateway is an external standalone device that provides cost effective communications or IP Telephony through the internet. It allows to connect two standard analog phone devices like home desktop phones, wireless phones, faxes or…
Read More »DNS NAPTR
Posted: November 27, 2003 DNS NAPTR resource records are documented in RFC 2915: This document describes a Domain Name System (DNS) resource record which specifies a regular expression based rewrite rule that, when applied to an existing string, will produce a new…
Read More »ICE: Interactive Connectivity Establishment
Posted: November 27, 2003 What is ICE? From draft-ietf-mmusic-ice-19.txt draft: This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of…
Read More »SER module nathelper
Posted: November 26, 2003 nathelper: This is a module to help with NAT traversal. In particular, it helps symmetric UAs that don’t advertise they are symmetric and are not able to determine their public address. fix_nated_contact: rewrites Contact header field with request’s…
Read More »Viseon VisiFone
Posted: November 26, 2003 The Viseon VisiFone is made by Leadtek. Price listed on the website is $599, but there is no indication of how to purchase one. From the website The VisiFone is the world’s first low-cost broadband videophone. It is…
Read More »Asterisk config modules.conf
Posted: November 22, 2003 Modules.conf configures which modules to load or not to load from the directory specified in asterisk.conf Sample config ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes ; ; noload => pbx_gtkconsole.so ;load =>…
Read More »SIP method ack
Posted: November 20, 2003 SIP implements a three-way handshake. The caller sends an INVITE The callee sends an 200 OK to accept the call The caller sends an ACK to indicate that the handshake is done and a call is going to…
Read More »Asterisk CDR csv handling
Posted: November 18, 2003 How a company handles CDR files from many Asterisk servers We have a cron job that moves master.csv to subdir/<timestamp>.csv and then tries to parse and delete all csvs in that directory. The reasons we did this were…
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