Author: VoIP Info
Asterisk bounty Generic Event Package Support
Posted: May 12, 2008 Bounty: Create Generic Support for SUBSCRIBE Event Packages From: John Dupuy <jdupuy AT socket.net>Bounty: 100 USDDate opened: May 12th, 2008 Description: I am offereing a $100 USD bounty to add generic support for SUBSCRIBE Event Packages that would…
Read More »AT-530P
Posted: May 12, 2008 IP Phone ：AT-530P Infineon Chipset Inside High performance IP phone with Infineon solution and PoE feature AT530P series IP phone is an internet based voice network phone terminal supporting power supply through Ethernet. AT530P series IP phone adopts…
Read More »IP08
Posted: May 9, 2008 8 FXO/FXS port IP PBX The IP08 is a complete Asterisk Appliance with four Dual port FXO or FXS module. It is an embedded Linux system with built-in SIP/IAX2 proxy server and NAT functions. It provides a solid,…
Read More »AT-510P
Posted: May 9, 2008 AT5-10P series IP phone is an internet based voice network phone terminal supporting power supply through Ethernet. AT-510P series IP phone adopts multiple voice control protocols and voice compression codec to directly convert analog voice into IP packet…
Read More »current ongoing development of chan_sccp-b
Posted: May 5, 2008 check out sourceforge http://sourceforge.net/projects/chan-sccp-b/ website: http://chan-sccp-b.sourceforge.net/ Channel Driver Download, Bugtracking, and Mailling list Current Development of chan_sccp-b on Sourceforge: Sample XMLDefault and SEP files Cisco 7914-HOWTO Current Development: SVN: svn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b chan-sccp-b Stable Asterisk 1.2 / 1.4…
Read More »VLAN – Virtual LAN for VoIP networks
Posted: April 30, 2008 Aim: Keep the phones working even when the data network is congested. The “Voice VLAN” is a special access port feature of Ethernet Switches which allows IP Phones to auto-configure and easily associate to a logically separate VLAN….
Read More »SafiWorkshop – Visual Call Flow Editor and Server for Asterisk
Posted: April 29, 2008 About SafiWorkshop and SafiServer SafiWorkshop is the first software offering for the Oregon based software company Safi Systems LLC. SafiWorkshop is a visual logic flow designer that allows Asterisk administrators to quickly create and deploy powerful IVR, auto-attendants,…
Read More »RFC4904
Posted: April 29, 2008 RFC 4904: Representing Trunk Groups in tel/sip Uniform Resource Identifiers (URIs) Abstract This document describes a standardized mechanism to convey trunk group parameters in sip and tel Uniform Resource Identifiers (URIs). An extension to the tel URI is…
Read More »Elistas networks Services Limited
Posted: April 29, 2008 Elistas Network Services Ltd – Telecommunications Company With over 5 years experience in the Telecommunications craft, Elistas Network Services Ltd is a progressive business, offering voip and carrier services to customers throughout the local area. Set-Up in 2005,…
Read More »Voip International Market
Posted: April 28, 2008 U.S. VoIP Market: The US market for VoIP advanced dramatically in 2006-2007, adding 3.8 million VoIP households in 2006, reports In-Stat: As a result, wholesale VoIP revenues grows quickly, as MSOs, Skype, and a myriad of new entrants…
Read More »Credit Card Dialplan for Asterisk
Posted: April 28, 2008 Here is a working model for asking the questions needed to process the creditcard. I have also added in this code something calledpaylib.py. This file is not complete and renders the code unable to process cards. This is…
Read More »Super Skype Gateway
Posted: April 26, 2008 The Best Solution to Take the advantages of Skype Unlimited Calls to Landlines Skype Unlimited Calls to Landlines from Skype.com Callfree Gateway 1600 Professional – CG1600PRO – Save Up to 160,000 Minutes per Month One of the best…
Read More »AT-510
Posted: April 26, 2008 Infineon Chipset Inside AT510 series IP phone is an internet based voice network phone terminal supporting power supply through Ethernet. AT510 series IP phone adopts multiple voice control protocols and voice compression codec to directly convert analog voice…
Read More »Toll Free Termination Providers
Posted: April 25, 2008 Toll-free termination are calls destined to 8YY destinations. These providers allow you to terminate toll-free calls from the US and Canada for free in some cases. If you have a large volume of calls to toll-free numbers, some…
Read More »Perl Add/Modify/Delete user script
Posted: April 24, 2008 i have created AddUser and EditUser scripts in perl for adding/modify/view/delete sip/voicemail entry in config files. Requirements: Asterisk::config CPAN module Features and working add/edit sip entry add/edit voicemail entry if required add/remove hint in extensions.conf if required. when…
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