VoIP Blog - Page 842

Compile Asterisk Ok, you have the source code to Asterisk sitting in your /usr/src/asterisk directory. Now what? cd /usr/src/asterisk make clean make make install Note: By default, Asterisk runs as the root user. This is a security liability….

Product discontinued? Website content missing as of 2009-02-24 See: http://www.myjabber.net The myJabberAE SIP soft phone integrates well with the myJabber IM. SLTS Communications myJabber License Agreement IMPORTANT NOTICE: The free download version of myJabber is intended for use…

IEEE (Institute of Electrical and Electronics Engineers) http://www.ieee.org/ From the IEEE website: Through its technical publishing, conferences and consensus-based standards activities, the IEEE produces 30 percent of the world’s published literature in electrical engineering, computers and control technology,…

Vovida.Org – Open Source Communications Vovida is a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments. Products offered by Vovida include: VOCAL a softswitch A SIP load balancing…

Making MVP210/410/810 and Asterisk working together. There’s a few parts envolved: 1. Initial configuration: sadly this has to be done with Multitech’s windows only program. Connects the modem cable to MVP810 and an available pc serial port, start…

Primus Canada offers a service called . Subscribers to this service get a real PSTN phone number that people can call (you get to pick which area code it’s in, and can even use your existing phone number)….

Linux QoS and Router Howto In this howto, I would like to demonstrate how to build a powerful router that supports QoS, stateful firewalling and many other features that linux is capable of. All using a spare computer…

Page Contents Example 1: June 05, 2004 October 6, 2004 Nov. 2, 2005 See also Zaptel PRI cards (T100P, E100P, T400P, TE405P, TE410P) can be configured for data (HDLC “family”) communication (WAN router), as well as hybrid configurating…

This variable should contain the number of the called party. Unfortunately it is broken for now and only contains the second part of the internal/temporary channel information (the part after the technology ie gs1-8b21 if the DIALEDPEERNAME was…

This page is intended to exchange experiences with Nikotel and their VoIP service. Feel free to add your own experience. If you are Nikotel staff, please be reminded that this is a community site where members of the…

regcontext regcontext=<context> If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us. The actual extension is the ‘regexten’ parameter of the registering peer…

GetGroupCount Synopsis GetGroupCount([groupname][@category]) Note: This command is deprecated in Asterisk v1.2. Instead use GROUP_COUNT([groupname][@category]) Set(GROUPCOUNT=${GROUP_COUNT()}) Description Calculates the group count for the specified group, or uses the current channel’s group if not specifed (and non-empty). Stores result in…

Zope Asterisk Objects This product allows Zope to connect to and manage an Asterisk PBX. Motivation Zope is an open-source web-application server based on the Python programming language. Zope provides a fully component oriented approach to building web-applications….