2N NETSTAR VoIP PBX
Posted: May 9, 2006 Advanced communication system 2N NETSTAR is is a communication system that integrates well-proven ISDN technologies with advanced GSM and VoIP communication networks. It works both like a classic digital switchboard and VoIP softswitch, which allows to utilise such…
Read More »Polycom 601 w/Expansion Modules
Posted: May 9, 2006 I have a Polycom 601 with two expansion modules. I’m running the latest firmware: 1.6.5..I am not using Asterisk..I am using a Broadsoft platform. I have two problems. The first is, I can’t get any of the directory…
Read More »Newman Ventures
Posted: May 9, 2006 Recently Posted Modules NVFaxDetect – Detects fax, talk, DMTF, and other tones on IAX, SIP, ZAP, and other channels NVBackgroundDetect – Detects fax, talk, DMTF, and other tones on IAX, SIP, ZAP, and other channels with background file…
Read More »AstLinux Development Environment
Posted: May 8, 2006 20 January 2010 Astlinux has now reached version 0.7. These voip-info.org pages are therefore extremely out-of-date. Please refer to the official Astlinux site: https://www.astlinux-project.org/ The AstLinux development environment is what was used to create AstLinux 0.4 The AstLinux…
Read More »Asterisk Native Sounds
Posted: May 8, 2006 Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729 licenses! The Asterisk Native Sounds are a collection of…
Read More »Cisco ATA 186 SIP and Asterisk – HowTo
Posted: May 8, 2006 ATA186 configuration. The ATA configuration is well described by Cisco you will find pointers on the following page Asterisk phone cisco ATA18x For a 5mn quick start do the following : Connect your ATA on the Ethernet, it…
Read More »Cisco ATA 186 MGCP and Asterisk – HowTo
Posted: May 8, 2006 ATA186 configuration. The ATA configuration is well described by Cisco and you will find pointers on the following pageAsterisk phone cisco ATA18x For a 5mn quick start do the following : Connect your ATA on the Ethernet, it…
Read More »Example Outbound Macro
Posted: May 8, 2006 Example Outbound Macro: This is the outbound Macro I use, it helps with routing calls to different destinations based on where the call is going to. ie certain international calls can be sent to a different carrier, or…
Read More »Asterisk bounty SIP answer supervision
Posted: May 7, 2006 Bounty USD $300Date 5/7/2006 I’am offering a $300 bounty and a bonus for a fix for asterisk to reconise the off hook data sent from my FXO gateway (VG 2400).This off hook data need to engage mysql to…
Read More »AMP-TelCommOne
Posted: May 7, 2006 The settings below can be used with AMP (Asterisk Management Portal) and Tel Comm One: Tested on version 1.10.010 IN TRUNKS: Add a SIP Trunk: Outbound Caller ID: (Blank) Maximum channels: (Blank) Outgoing Dial Rules Dial Rules: $PIN$+NXXNXXXXXX…
Read More »AsteriskatHomeTelCommOne
Posted: May 7, 2006 The settings below can be used with Asterisk@Home 1.3 & 1.5 / AMP (Asterisk Management Portal) and Tel Comm One: IN TRUNKS:Add a SIP Trunk: Outbound Caller ID: (Blank)Maximum channels: (Blank) Outgoing Dial RulesDial Rules:%PIN%+NXXNXXXXXX%PIN%+1NXXNXXXXXX%PIN%+011. Outgoing Dial Prefix:…
Read More »VoicePulse Connect for Asterisk
Posted: May 5, 2006 VoicePulse FIVE Next Generation Phone Service. Connect for Asterisk has been replaced with VoicePulse FIVE our fifth generation VoIP platform. FIVE is a completely rewritten platform designed from the ground up to provide Internet telephony in the most…
Read More »SipLinks WebPhone
Posted: May 5, 2006 2009-02-24 – Product Discontinued? Website no longer found. 2009-10-30 – Similar product can be found on http://www.wallusoft.de – WebPhone (click on the test button for a demo) 2009-11-12 – Similar product can be found on http://www.doddlephone.com – (Doddle…
Read More »SipLinks
Posted: May 5, 2006 Weblinks for SIP WebPhone. SipLinks enables CUSTOMIZABLE WebPhones for your website. We provide you with a web link that can be placed anywhere on your web site. It enables your site visitors to make SIP based phone calls…
Read More »Asterisk sip domain
Posted: May 5, 2006 sip.conf domain= With the domain statement you may handle more than one domain on an asterisk server. In section [general] you may add domain=<mydomain>,<context> Calls to <mydomain> will go to context [<context>], overriding (!) the general and peer…
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