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)…
Read More »Siproxd
Posted: March 23, 2004 http://siproxd.sourceforge.net/ From the website: Siprox is an proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections possible via…
Read More »Asterisk cmd Eval
Posted: March 22, 2004 Synopsis: Eval(newvar=somestring) Description: Eval(newvar=somestring) Normally Asterisk evaluates variables inline. But what if you want to store variable offsets in a database, to be evaluated later? Eval is the answer, by allowing a string to be evaluated twice in…
Read More »Tedas
Posted: March 20, 2004 From website: The most important product in the spectrum is the Phoneware IP telephony server. For that, we offer you a large selection of desktop solutions which suit to every workplace. Add-ons, a fax function and voice mailbox…
Read More »HelloWorld
Posted: March 20, 2004 MIOCO hello world! products comprise a range of software for PC-based telephony. Professional users make mobile calls with their notebook, from their home PC or from their workstation within the company to contacts anywhere in the world via…
Read More »IETF
Posted: March 11, 2004 Internet Engineering Task Force Standards body for Internet protocols and services. The keeper of RFCs. http://www.ietf.org/ IETF Working Groups related to VOIP >SIP Working group The Session Initiation Protocol (SIP) working group is chartered to continue the…
Read More »Asterisk last number repeat
Posted: March 9, 2004 Asterisk Last Number Repeat If you use this setup a phone can dial *5 to repeat the last number you dialed (from that callerid). The configuration is based upon a few steps: Step 1: Separate the dialling context…
Read More »Dialexia
Posted: March 7, 2004 www.dialexia.com From website: Dialexia Communications is an award-winning Canadian telecommunications software & services company specialized in the development of innovative VoIP business phone solutions for corporations, small businesses & telephony resellers alike. The company also provides custom-built PBX…
Read More »sipd
Posted: March 7, 2004 From website – March 2004 sipd is a high performance, scalable SIP (Session Initiation Protocol) proxy and location server written in C. It is thread-based and highly modular, and its location database and transaction layers are based on…
Read More »Linphone
Posted: March 7, 2004 http://www.linphone.org/ (French) Linphone is an open source (or free software) SIP Softphone, with support for several codecs. It runs using a gtk interface linux and windows. Besides the graphical interface, some command line interfaces are available, called linphonec…
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