VoIP Blog - Page 837

PortaOne’s Radius client for Asterisk Overview This GPL project brings Radius AAA capabilities to Asterisk PBX. It is written entirely in Perl, so it is 100% portable and easy to understand and customize! RADIUS attributes are used as…

Asterisk call through This is a simple call through example, after calling the system (extension 500), enter a phonenumber and the system dials it after pressing ‘#’. You can reenter the number after pressing ‘*’. After the “call…

Pulse Dialing on Zap Channels Since 1.0.1 Asterisk has pulse dialing support for zapata channels. Just specify pulsedial=yes in zapata.conf Workaround for Incorrect Pulse Decoding If you are using FXS ports on a TDM400 card and are having…

  Dial out routines to find next best lowest cost channel [macro-dial-pstn-sip-iax] ; ; ; This macro tries the PSTN circuit first, then ; fails over to our SIP provider and uses IAX ; as a last resort….

GSM stands for Global System for Mobile Communications. It is the dominant mobile phone communication protocol in Europe and also popular elsewhere. The name is also used to describe some of the voice codecs used in the GSM…

G.722 is a high bit rate (48/56/64Kbps) ITU standard codec. ITU G series recommendations There was long thread on the one of the Asterisk Mailing Lists recently about the G.722 codec.. Some of the more interesting messages: Andrew…

Digium Certified Asterisk Professional dCAP, Digium Certified Asterisk Professional, is a verification of your knowledge of Asterisk gained in the training class. The certification is personal and is given for a version of Asterisk. The current dCAP exam…

In 2007, Polycom acquired Spectralink, the industry leader in Voice-over-WiFi technology. This acquisition followed Polycom’s earlier purchase of Kirk Wireless, a leader in DECT wireless equipment manufacturing. These acquisitions enabled Polycom to leapfrog into a leading roll in…

Azatel designs, manufactures and integrates SIP & MGCP related products to provide our partners with highly advanced Analogue Terminal Adapters (ATA) and cost effective SIP solutions. AZACALL200 Our flagship model, the Azacall200, has gained significant industry recognition from…

An editor for editing binary files. There contents for the file are shown in Hexidecimal, although binary, decimal, octal, and ASCII display modes may also be available. Windows Hex Editors wxHexEditor http://wxhexeditor.sourceforge.net Opensource – GPL license Native Win32…

This page documents how you configure a Cisco 12SP+/30VIP phone with Asterisk. The Cisco 12SP+/30VIP IP Phones are EOL and they are not supported by Cisco anymore. Cisco states that they do not have any listings for this…

E1 is a the European equivalent of a T1 but runs at 2.048Mbps and has 32 x 64Kbps channels DS0s where 30 is used for voice and two are used for syncronization and signaling. Alarms that can occur…

Bounty Paid: 12/8/2004 to Theo Zourzouvillys Originally documented in https://issues.asterisk.org/jira/browse/ASTERISK-2326. Currently if we have Asterisk SIP channel driver binding to all interfaces, and eth0 has many subnets attached to it (a primary 10.1.200.1, and then alias interfaces eth0:1…