Asterisk SIP Channels
Posted: August 9, 2005 The SIP Channel Module enables Asterisk to communicate via VoIP with SIP telephones and exchanges. Asterisk is able to act as: a SIP client: This means that Asterisk registers as a client to another SIP server and receives…
Read More »Asterisk cmd BristuffDevstate
Posted: August 9, 2005 app_devstate (bristuff) Application for sending device state messages. Mainly used for lighting the LEDs on the snoms. Generate a device state change event (inuse, busy, ringing …) devstate(device,state) Details Devstate() is part of the Bristuff patches (0.2.0-RC7i/j and…
Read More »Leadtek Research
Posted: August 9, 2005 Leadtek Research Inc. Company Website Manufacturer of ATAs & Videophones SIP ATA products: Leadtek BVA8055 1 FXS port Leadtek BVA8051S (also sold as SIPphone Call-in-One) 1 FXS port & 1 FXO port Leadtek BVA8052D 2 FXS port Leadtek…
Read More »Asterisk Manager API Action QueueRemove
Posted: August 9, 2005 Action: QueueRemove Parameters: Queue: Existing queue to remove member from Interface: Member interface (sip/1000, zap/1-1, etc)? Example: Action: QueueRemove Queue: queue1 Interface: sip/2600 See also Asterisk Manager API Action QueueAdd Asterisk manager API Asterisk config queues.conf Asterisk cmd…
Read More »Asterisk Linux Mandrake
Posted: August 8, 2005 Asterisk on Mandrake Linux General Since Mandrake 10.1, asterisk is available in contrib, with the zaptel tools available in the zaptel-utils package. The zaptel drivers are available in contrib (as a dkms package dkms-zaptel) since Mandriva 2005. Installing…
Read More »Asterisk Linksys WRT54G
Posted: August 8, 2005 Asterisk on WRT54G and compatible routers Brian Capouch did demos of Asterisk running on a Linksys WRT54G at the Spring 2005 VON in San Jose. He teaches a course on VOIP and has each student use Asterisk on…
Read More »SER example MySQL
Posted: August 8, 2005 SIP Express Router example: SER working with MySQL $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ simple quick-start config script ———– global configuration parameters ———————— debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no #…
Read More »Converting RTP to audio
Posted: August 8, 2005 For debugging and troubleshooting it is often desirable to convert a stream of captured RTP packets to playable audio (e.g. a WAV file). A major difficulty in doing this is decoding the many different types of Codecs that…
Read More »Asterisk Inphonex
Posted: August 8, 2005 This is what i had to do to setup my $7.95 Miami inbound DID with InPhonex sign up on their site for pay as you go outbound service $9.95 + $1.00 tax After signing up you will have…
Read More »Asterisk T.38 Bounty
Posted: August 8, 2005 Bounty for SIP T.38 fax gateway support in Asterisk I need to do as follows: Ordinary fax machine connects to SIP/T.38 enabled ATA that calls Asterisk server (1) Asterisk server (1) forwards the call to Asterisk server (2)…
Read More »MSN PHP
Posted: August 8, 2005 (Work in progress – will update as the code becomes more stable) I had a neat idea to have asterisk send me a MSN message based on an Asterisk event. I was able to have asterisk execute an…
Read More »VoIP and Wiki
Posted: August 8, 2005 Does a combination of VoIP and Wiki make sense? On http://www.emacswiki.org/cgi-bin/community/VoiceOverInternetProtocol we started thinking on using telephony in the wiki process. It would be interesting to hear other opinions on the issue, not only about the social but…
Read More »Asterisk Cisco 79XX XML Services
Posted: August 7, 2005 The Cisco 79XX Phones include a mini-browser that allows the phone to interact with specially designed web services. These services can be developed as a CGI script on an HTTP server which outputs content using the XML syntax…
Read More »AsteriskAtHomeViaTalk
Posted: August 7, 2005 FreePBX / Asterisk@Home Config for ViaTalk dsherron FreePBX / Asterisk@Home working config for ViaTalk tested with A@H 1.3 and FreePBX 2.1.3 Replace all sections in <brackets> with information from your setup email. Do NOT leave the brackets. If…
Read More »VoIP User Groups csa
Posted: August 6, 2005 Asterisk Colombia Interesados en participar en el grupo Asterisk Colombia, ya tenemos el sitio AsteriskColombia.org con el apoyo de GECKO, la principal firma en Colombia de soluciones basadas en Asterisk y representante autorizado de Digium. Pueden escribir a:…
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