Author: VoIP Info
Analog Telephone Adapters
Posted: September 21, 2005 Important notice to product manufacturers, resellers and other people adding to this page If you want to add your company’s products, please read the Posting Guidelines for Promoting Products and Services The lists shall be in alphabetical order;…
Read More »Asterisk Cisco CallManager Voicemail Integration
Posted: September 21, 2005 This is my first wiki entry and needs to be tidied up a bit. I’ll do that when I get around to it, my main priority for now was getting this information online as it seems to be…
Read More »Asterisk tips Wake-Up Call PHP
Posted: September 21, 2005 Set a Wake Up call Features: Schedule or delete a single wake-up call No means to edit an entry No CRON job or daemon needed, scheduling works by .call file timestamp Ubuntu 13.10 – After upgrading to Ubuntu…
Read More »Asterisk phones
Posted: September 21, 2005 Asterisk Configuration Notes for Specific Phones These pages document how to configure Asterisk for specific telephony devices. Asterisk Configuration Notes for Specific Hard Phones and ATAs Aastra / Sayson phones: Asterisk and Aastra Phones Atcom IP Phone:AT-510/AT530:iax2/sip support…
Read More »IXC
Posted: September 20, 2005 IXC offers VoIP solution, designed to handle contemporary VoIP carrier’s or service provider requirements. Portfolio of the company constitutes Billing Retail, Billing Wholesale, IXC Softswitch and IXC Autotester. * Main development directions Softswitch, VoIP solution with build-in web-based…
Read More »PERL-EMAIL-AGI
Posted: September 20, 2005 !/usr/bin/perl Added by Suretec Systems Ltd. – http://www.suretecsystems.com – 28/04/2006 use warnings; use strict; use Asterisk::AGI; use File::Basename; use Mail::IMAPClient; use MIME::Parser; use Digest::MD5 qw(md5_hex); use POSIX; my $DEBUG = 1; my $T2WP = “/usr/bin/”; # path…
Read More »Monitor setup sample
Posted: September 20, 2005 The following only works with CVS head as of 2004-06-03. It allows a user to call out by dialing 8, then the number. Any calls dialed out using 8 are recorded. The “m” option on the 2nd set…
Read More »STUN
Posted: September 20, 2005 STUN stands for Session Traversal Utilities used for NAT (Network Address Translation). It is a server-client protocol for assisting devices behind a NAT firewall or router with their packet routing. RFC 5389 redefines the term STUN as ‘Session…
Read More »QueueMetrics
Posted: September 20, 2005 QueueMetrics is an inbound and outbound call center analysis, monitoring and reporting software application. The current version is 17.06 as of Jun 21, 2017. QueueMetrics is available as an on-premise or hosted solution. It offers over 150 quantitative…
Read More »SIP Video Phones
Posted: September 20, 2005 Hardware Phones: TANDBERG E20 – SIP VOIP and Video Phone Grandstream GXV3000 – Cheap H.264 SIP Videophone Grandstream GXV3240 IP Multimedia Phone Grandstream GXV3275 SIP Videophone – Voice and/or video H.264/H.263/H.263+ and embedded Skype, browsing etc too. fanvil…
Read More »ShadyDial
Posted: September 20, 2005 http://shadydial.sourceforge.net/ Predictive Dialer for Asterisk The link above at sourceforge is quite old, but does get you a feel for the product. The database backend is still PostgreSQL and it uses stock Asterisk parts, along with some freely…
Read More »Asterisk func callerid
Posted: September 19, 2005 Synopsis: CALLERID(<item>) Description: CALLERID(<item>) The allowed values for <item> are: name: alphanumeric string num or number: number at which caller prefers to be called back (digits only in the POTS/ISDN world) all: a Caller*ID string with the number…
Read More »GizmoProject
Posted: September 19, 2005 Gizmo Project is a closed-source software phone developed by SIPphone. It is a VoIP phone, made specifically for the SIPPhone network. It complies with Open Standards, directly peers with several other VOIP providers, and will directly dial sip…
Read More »Roby Technology Company
Posted: September 19, 2005 Showroom A SIP gateway ,with 1 FXS and 1 LAN.Nice voice quality and easy to install ! IP phones: support IAX2, SIP ,H.323,MGCP,Net2phone. A USB skype phone with a LCD screen. Company Profile China Roby Technology Co.,Ltd. was…
Read More »Low Bandwidth VOIP
Posted: September 19, 2005 Although voice quality may suffer, VOIP can be used over a dial-up Internet or other low bandwidth connection. Some users have reported being able to use the Vonage service over dial-up, although Vonage doesn’t officially support this. Hints…
Read More »
Phone System 