Author: VoIP Info
Asterisk tips renumber extens
Posted: January 4, 2005 Adding a new rule in the middle of extensions.conf can be painful if you need to renumber the priorites of things below it. I wrote a little perl script to renumbers the priorities. It looks at rules in…
Read More »Asterisk config codecs.conf
Posted: January 4, 2005 Configuration file for asterisk codecs. Example for Asterisk 1.2: See also bug/patch 2536 [speex] ;0-10 quality => 3 ;0-10 complexity => 4 ; true / false enhancement => true ; true / false vad => false ; true…
Read More »VOIP Payphones
Posted: December 31, 2004 This page now has VoIP Payphones from several vendors. I have no experience with any of these companies, nor their products. They are just the result of my search for a VoIP payphone. Vector Technology Corp. Provider of…
Read More »Freebusy
Posted: December 31, 2004 Freebusy Concept, check an Microsoft Exchange server for freebusy information against a calendar. This could be used for both user calendars – to send straight to voicemail (auto do-not-disturb) or for resource scheduling – such as maintaining a…
Read More »Asterisk x100p echotraining
Posted: December 31, 2004 Echo Cancellation on the Wildcard X100P If you are experiencing echo on the Digium Wildcard X100P, you can configure the Zap Channel Module to do echo cancellation training. Use the echotraining option in the Zap Channel Module’s configuration…
Read More »Asterisk Tips IVR Menu
Posted: December 30, 2004 Implementing a Simple ‘Push-1, Push-2’ Menu Structure The key to creating this menu is to create an Extension (defined as 205 below) to record your menu prompts. This will put the sound file in /tmp/asterisk-recording.gsm. You’ll have to…
Read More »RTP
Posted: December 30, 2004 RTP opens two ports for communication. One for the media stream (an even port number) and one for control (QoS feedback and media control) – RTCP. The port numbers are not hard defined, it depends very much upon…
Read More »Asterisk Letting SIP clients connect directly
Posted: December 30, 2004 Asterisk by default connects all media streams through asterisk to be able to connect various protocols and media to each other. If you have two SIP phones, the media path can be connected directly between the phones without…
Read More »Asterisk Understanding the source code
Posted: December 29, 2004 When studying the Asterisk source code the following suggestions may prove useful: Running CTAGS on the Asterisk source code then loading the Asterisk source code into an editor that supports CTAGS will create an environment where you can…
Read More »Asterisk cmd ParkedCall
Posted: December 28, 2004 Synopsis Answer a parked call Description ParkedCall(exten) Used to connect to a parked call. This Application is always registered internally and does not need to be explicitly added into the dialplan, although you should include the ‘parkedcalls’ context….
Read More »CDR mediation
Posted: December 27, 2004 CDR mediation CDR mediation is intermediary process to billing which follows CDR collection. This is necessary to make sure calls are billed to the right entity and based on the right tariffs. CDR mediation consists of several processing…
Read More »sems
Posted: December 25, 2004 SEMS: SIP Express Media Server Introduction: SEMS is a free, high performance, extensible media server for SIP (RFC3261) based VoIP services. It is intended to complement proxy/registrar servers in VoIP networks for all applications where server- side processing…
Read More »PortaOne Radius auth
Posted: December 24, 2004 PortaOne’s Radius client for Asterisk Overview This GPL project brings Radius AAA capabilities to Asterisk PBX. It is written entirely in Perl, so it is 100% portable and easy to understand and customize! RADIUS attributes are used as…
Read More »Asterisk tips call through
Posted: December 24, 2004 Asterisk call through This is a simple call through example, after calling the system (extension 500), enter a phonenumber and the system dials it after pressing ‘#’. You can reenter the number after pressing ‘*’. After the “call…
Read More »Asterisk zaptel pulse dialing
Posted: December 24, 2004 Pulse Dialing on Zap Channels Since 1.0.1 Asterisk has pulse dialing support for zapata channels. Just specify pulsedial=yes in zapata.conf Workaround for Incorrect Pulse Decoding If you are using FXS ports on a TDM400 card and are having…
Read More »
Phone System 