Author: VoIP Info
check_asterisk
Posted: July 28, 2005 Asterisk SIP Channel Nagios Plugin Special thanks to Nils Ohlmeier for writing SIPSAK and Alexander Mayrhofer for writing the original version of this plugin for SER. The Nagios Check command can be something like: ‘check_asterisk’ Asterisk command definition…
Read More »Configuring GSM VoIP gateways with Cisco Call Manager
Posted: July 28, 2005 Configuring GSM VoIP gateways with Cisco Call Manager The present document is a step-by-step guide for configuring the GSM VoIP gateways 2N VoiceBlue Lite (VBL), 2N VoiceBlue Enterprise (VBE) and Cisco Call Manager (CCM) software IP PBX. It…
Read More »Asterisk SCCP channels
Posted: July 28, 2005 SCCP channel drivers in Asterisk There are 2 variants of SCCP channel drivers which exist for the Asterisk PBX: Supplied driver with Asterisk chan_skinny or Asterisk channel skinny The “original” Driver written by Jeremy McNamara & Florian Overkamp…
Read More »MozPhone
Posted: July 28, 2005 Mozphone has been renamed MozIAX and now has a home page at mozdev.org This is a stub page to contain information about how to configure MozPhone to work with Asterisk. Usage Choose MozPhone in Firefox Tools menu. You…
Read More »How to start a Clec
Posted: July 27, 2005 How to become a CLEC(Well ok, just a start) Extracted with thanks from http://www.robotics.net/papers/clechowto.html By: Nathan Stratton nathan@robotics.netThis is a draft document, please email me with changes or suggestion nathan@robotics.net Introduction The Telecommunications Act if 1996 (Act) opened…
Read More »RFC 3903
Posted: July 27, 2005 Session Initiation Protocol (SIP) Extension for Event State Publication Abstract This document describes an extension to the Session Initiation Protocol (SIP) for publishing event state used within the SIP Events framework. The first application of this extension…
Read More »Asterisk Cisco CallManager Integration
Posted: July 27, 2005 Why integrate Cisco CallManager and Asterisk? Features: Asterisk provides features that CallManager by itself does not. Migration: Allow a gradual migration from a closed source PBX to open source PBX. There are two ways to accomplish this: Using…
Read More »ENUM – The bridge between the switched telephony network and the Internet
Posted: July 27, 2005 Page Contents News IETF Documents Selected List of Published RFCs Internet Drafts in the Pipeline ENUM Experts Various Information on ENUM Current ENUM deployment status in various countries Software support Record syntax Organizations involved with ENUM Links Articles…
Read More »STUN clients
Posted: July 27, 2005 Known STUN clients: xten softphones: X-Lite, X-PRO, eyeBeam Linux client on http://sourceforge.net/projects/stun/ SNOM Phones Cisco Ata v 3.0 firmware pbxnsip PBX for trunks Grandstream Budgetone and Handytone Hotsip Active Client Sipura All SPA Products Leadtek Research BVA8051, BVA8052,…
Read More »Asterisk mark2 echo canceller
Posted: July 27, 2005 Echo cancellation in Asterisk — ECHO_CAN_MARK2 based on code available from Mississippi State Univerisity Paramters (in mec2_const.h) DEFAULT_BETA1_I 1/(compromise stepsize constant, beta1); higher value means a slower update DEFAULT_SIGMA_LY_I -log2(filter constant) for reference power estimate; higher value…
Read More »Asterisk config zaptel.conf
Posted: July 27, 2005 Configuration File /etc/zaptel.conf PageNeedsRevision: information in this page is quite obsolete and not well-edited. As it is it is not useful enough. The zaptel.conf file is where you configure the TDM-specific interface parameters required by your Zaptel card(s)….
Read More »GEDAM_Europe
Posted: July 26, 2005 GEDAM Europe SRL, Torino GEDAM Europe The European Arm of GEDAM Advanced Communications in New Zealand and Partner of SineApps (The producer of the Daily Asterisk News has finally gone public. The main base of operations for Europe…
Read More »Asterisk cmd ChanSpy
Posted: July 26, 2005 ChanSpy Synopsis Listen in on a call, or whisper into a conversation. Useful in a call center to monitor agents on the phone. Description This adds the ability to spy on any bridged call, this includes VoIP-only calls…
Read More »SIP SS7 gateways
Posted: July 26, 2005 SIP <-> SS7 Gateways SS7 (or C7) is the main signalling for PSTN interconnections. How do we interface between a SS7 network and a SIP network? SIP to SS7 Gateway from Terratel The signaling converter allows to combine…
Read More »FreePBX
Posted: July 25, 2005 FreePBX is a web application. If you’ve looked into Asterisk, you know that it doesn’t come with any “built-in” programming. You can’t plug a phone into it and make it work without editing configuration files, writing dialplans, and…
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