FLEXCEL – GATEWAY GSM VOIP VP-100Br
Posted: August 26, 2008 Gateway VoIP GSM VP-100Br1SIM, 1LAN, 1FXS+PSTN, SMSQuad band 900/800/850/1900MHzCompatàvel com Asterisk, SIP, STUN, ENUM Faà§a chamadas do Seu Celular para qualquer lugar Via VoipDo seu Celular em qualquer lugar, a qualquer hora! . Caracteràsticas:. 1 porta FXS para…
Read More »UM Labs Ltd
Posted: August 19, 2008 Security for VoIP and Unified Messaging! The focus of UM Labs is the delivery of a range of high-quality VoIP perimeter security products. A series of product releases are planned through 2008 to address all markets from Branch…
Read More »Asterisk Integration with Altigen
Posted: August 13, 2008 Integration of Asterisk system with Altigen Equipment: Legacy Altigen system in a 3 chassis design, supporting over 340 extensions Installed are TritonAL12EXT-3 cards for connection of up to 12 analog phones each. These boards are for homing extensions….
Read More »Asterisk with Altigen
Posted: August 13, 2008 Integration of Asterisk system with Altigen THERE IS A MORE COMPLETE VERSION OF THIS HERE: Asterisk Integration with Altigen Equipment:– Legacy Altigen system in a 3 chassis design, supporting over 340 extensions– Installed are TritonAL12EXT-3 cards for connection…
Read More »Asterisk sip limitonpeers
Posted: August 13, 2008 Typically needs to be set to ‘yes’ in Asterisk 1.4 or 1.6.0.x in order for hints/sip subscriptions/BLF to work as expected. note: it is limitonpeer, without s! Description Default is ‘no’; this setting applies to the [general] section…
Read More »Adhearsion
Posted: August 11, 2008 Project Website: adhearsion.com Documentation: adhearsion.com/docs API: adhearsion.com/api Community: adhearsion.com/community Latest Release: 2.1.0 released August 7th, 2012 (this wiki page last updated August 24th, 2012) Adhearsion is a full-featured framework for the development of applications which interact with or…
Read More »Interwork Technologies
Posted: August 11, 2008 Specialty Distributor for Voice Solutions! Interwork Technologies Inc. is a Value-Added Distributor of unified communications (VoIP) and IT Security Solutions. Interwork’s expertise is unmatched in the North American channel, providing solutions comprised of the very best in complementary…
Read More »SMS callback for Asterisk
Posted: August 11, 2008 This article was originally published in 1telecom.net SMS Callback for Asterisk PBX SMS callback allows you to eliminate DTMF detection problems and inconvenience when using a callback solution. In this section, I will show a generic example for…
Read More »Fee Announcement from Free World Dialup
Posted: August 6, 2008 Dear FWD’er: Free World Dialup is implementing a $30 USD annual membership fee during the month of August after 12 years as a free service. A startup taking over a decade to reach conviction about a business model…
Read More »Asterisk func shared
Posted: August 6, 2008 Synopsis Gets or sets the shared variable specified Introduced with Asterisk 1.6.1 Description SHARED(<varname>[,<channel>]) Implements a shared variable area, in which you may share variables between channels. If channel is unspecified, defaults to the current channel. Note that…
Read More »ASTELCENTER
Posted: August 6, 2008 ASTELCENTER is a PHP/MYSQL web solution development by ANTARTEL, which allows you to manage and supervise your ASTERISK PBX; generating reports and information in real time so clear and concrete; allows you to generate statistics and graphs: incoming…
Read More »USKY Skype Gateway UBS-200
Posted: August 6, 2008 Perfect choice for SMBs to build up Skype IP Tel-Platform Small and medium sized business can now build up Skype communication platform for company and enjoy free(Skype to Skype) and low(SkypeOut, Skype Unlimited Call Service) communication cost…
Read More »About OpenSIPS
Posted: August 4, 2008 OpenSIPS (Open SIP Server) (former OpenSER) is an Open Source SIP proxy/server for voice, video, IM, presence, and any other SIP extensions. OpenSIPS is a multi-functional, multi-purpose signaling SIP server – it can act as SIP Router/Switch, SIP…
Read More »chan_dahdi.conf
Posted: August 3, 2008 ; ; DAHDI telephony ; ; Configuration file ; ; You need to restart Asterisk to re-configure the DAHDI channel ; CLI> reload chan_dahdi.so ; will reload the configuration file, ; but not all configuration options are ;…
Read More »AACC
Posted: August 3, 2008 AACC (short for Asterisk Advanced Call Center) is an underway project which aims to create a full CTI solution for the Asterisk PBX. It is written in Java, which makes it operating system independant. It will feature: Inbound…
Read More »
Phone System 