VoIP Blog - Page 806
SER faq authenticate
Posted: July 30, 2005 Question In debugging yet another SIP hardware device I found that it was not authenticating after an INVITE. That is, the device sends me ‘INVITE’, I say ‘401‘, it says ‘SIP method ack‘, but I never hear from…
Read More »NVFaxEmail
Posted: July 29, 2005 See also NewmanTelOnAsterisk14 – Instructions to modify Newman Telecom modules for use on Asterisk 1.4.0 NVFaxDetect – Detects fax, talk, DMTF, and other tones on IAX, SIP, ZAP, and other channels NVLineDetect – Detects answer, busy, congestion, dialtone,…
Read More »AU-600
Posted: July 29, 2005 The AU-600 is a USB device which provides a traditional PSTN FXO and FXS interface to a Windows machine running the Skype VoIP client software. It allows users to use a regular analog telephone as the audio device…
Read More »ISDN Features
Posted: July 28, 2005 Common ISDN Supplemantary Services CLIP: Calling Line Identification Presentation CLIP – no screening: CLIP can be modified by the caller out of the bounds of it’s assigned number space CLIR: Calling Line Identification Restriction. CLIP is supressed. COLP:…
Read More »PRI
Posted: July 28, 2005 ISDN connections are delivered in two variants, PRI – Primary Rate Interface and BRI – Basic Rate Interface. PRI is the standard for connections to offices. It’s based on a T1 line in the US, and E1 line…
Read More »ISDN
Posted: July 28, 2005 ISDN (Integrated Services Digital Network) A series (I-series) of ITU standards for digital telephone service. ISDN is usually provided in two variants: BRI: Basic Rate Interface – BRI was never very popular but does still exist in North…
Read More »ITU
Posted: July 28, 2005 http://www.itu.int/ The International Telecommunications Union, in Geneva, Switzerland is an international organization within the United Nations System that issues ‘Recommendations’ for global telecom networks and services. Examples: E Series: Network Operations H Series: Audiovisual and Multimedia systems –…
Read More »RFC3372
Posted: July 28, 2005 RFC 3372: Session Initiation Protocol for Telephones Abstract The popularity of gateways that interwork between the PSTN (Public Switched Telephone Network) and SIP networks has motivated the publication of a set of common practices that can assure consistent…
Read More »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 »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 »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 »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)….
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