Author: VoIP Info
Asterisk bounty MGCP Media gateway
Posted: April 10, 2004 Maintainer: slepp Date opened: march 28, 2004 Description: The chan_mgcp appears completely designed to operate in CallAgent mode. We need it to operate in Media Gateway mode as well. Requirements: Register to MGCP Call Agent (via RSIP) Handle…
Read More »Asterisk presence jtodd
Posted: April 9, 2004 Work in progress, brainstorm – NO DOCUMENTATION WHATSOEVER John Todds outline John Todd: My idea for AIM/Jabber/Yahoo integration is below. Comments and/or programmers are welcome to have at it, and to expand on my ideas. I have mentioned…
Read More »SIP DTMF Signalling
Posted: April 8, 2004 Implementations of the Session Initiation Protocol (SIP) commonly use four methods for signaling digits between user agent servers and user agent clients: DTMF audio tones in the RTP stream (inband). This requires the use of lossless coders, such…
Read More »SER module auth
Posted: April 8, 2004 This is a generic module that itself doesn’t provide all functions necessary for authentication, but provides functions that are needed by all other authentication related modules (so called authentication backends). We decided to break the authentication code into…
Read More »Asterisk sip client SER
Posted: April 7, 2004 Connecting from Asterisk over NAT to IPtel.org Jan Janak of iptel.org writes: I am using [email protected], all the SIP traffic will be sent to iptel.org proxy and the proxy will take care of NAT traversal. Currently I forward…
Read More »Asterisk WAMI
Posted: April 6, 2004 Windows Asterisk Management Interface (note: this software does not seem to exist, and the web page linked below is a cyber squatter link farm. If you are the author, please update this page. – March 6, 2007) Latest…
Read More »Asterisk indications New Zealand
Posted: April 6, 2004 Asterisk indications for New Zealand Insert this into indications.conf and set country to NZ. asterisk/configs/indications.conf.sample contains a newer version of this data. [nz] description = New Zealand ring cadence = 400,200,400,2000 dial = 400 busy = 400/375,0/375 ring…
Read More »Asterisk man page
Posted: April 6, 2004 This is work in progress to create an Asterisk man page NAME asterisk – an Open Source multiprotocol PBX SYNOPSIS asterisk [-r] add syntax here DESCRIPTION Asterisk is a complete PBX in software. It runs on Linux and…
Read More »Asterisk Request Tracker Integration
Posted: April 5, 2004 How To integrate Asterisk with Request Tracker – a helpdesk/NOC Ticketing system. This helps you to create tickets in request tracker based on incoming calls. Request tracker handles web and email requests. The actual voice will be attached…
Read More »ICUC
Posted: April 1, 2004 ICUC ICUC has started a service to allow VOIP providers to query their serivce and find out if a call to a given phone number can be delivered via VOIP. Providers may both advertise the numbers they can…
Read More »Asterisk Manager Interface Debugging
Posted: March 31, 2004 I have recently noticed that the “Action: Originate” options in asterisk 1.0 Stable CVS has changed sometime between 2/23 and 3/18. It is now required to supply a Priority. Here are 2 tips were suggested when using the…
Read More »CallUK
Posted: March 30, 2004 In 1997, Telco Electronics formed CALL UK as its dedicated telecommunications provider. The company awarded its PTO licence in 1998, delivers niche managed services and network solutions aimed at specialist user groups, including educational institutions, international organisations and…
Read More »MailVisionNAT
Posted: March 29, 2004 MailVision NAT / Firewall Traversal Solutions: SERVER STUN Server TURN Server ( latest draft ) Outoing SIP Proxy Server Media Relay Server SIP B2BUA Server For NAT/Firewall Traversal CLIENT STUN / TURN Client Library ( Java and ANSI…
Read More »Asterisk CLI prompt
Posted: March 26, 2004 Changing the CLI Prompt The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable that you set from the Unix shell before starting the Asterisk CLI (not the server). You may include the following variables, that will…
Read More »Magrathea
Posted: March 24, 2004 Magrathea Telecommunications Magrathea provides wholesale IAX and SIP based call origination and termination services to other carriers and technical partners. Based in London, England. They can provide a range of UK based numbering to terminate on your Asterisk(IAX)…
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