VoIP Blog - Page 836
Asterisk sip language
This configuration option for a SIP account in Asterisk specifies language setting to be used for this client. By using this setting, you may get localized voice prompts in different languages for different users. Country settings Country setting…
Read More »DUNDi peers tier3 us
The original Tier 1 and Tier 2 Peers are listed at http://www.dundi.com/members.html Please use the peer listing format to list your information. Use the following links to locate a Tier 3 peer by State: Florida Illinois Iowa Minnesota…
Read More »ASTCC Bounty
Bounties to be paid for ASTCC functionality If you really want something added to ASTCC, but can’t do it yourself and do not want to contract an Asterisk consultants or Digium to do it, there’s always an opportunity…
Read More »Asterisk DUNDi Call Routing
How to use DUNDi to distribute calls among the DUNDi Peering Network General DUNDi Information Quote from http://www.dundi.info/: “DUNDi™ is a peer to peer system for locating Internet gateways to telephony services. Unlike traditional centralized services (such as…
Read More »REN
REN = Ringer Equivalency Number A rating for FXS ports that indicates how many phones can be simultaneously connected to the port. In days past, the old mechanical ringer telephones had REN ratings for 1.0 each, and there…
Read More »BYOD
Bring Your Own Device – BYOD The practice of some VOIP Service Providers to allow you to supply your own equipment. Other providers insist on supplying equipment, which usually is hobbled so it only works with one service…
Read More »Asterisk WhoIsIt
Who Is It Facilities to Announce incoming callers over the computer speakers I’ve written a small program to play announcements over the computer’s speakers when a call comes in, based on the CID of the caller. This can…
Read More »VOIP Payphones
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
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
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 Letting SIP clients connect directly
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
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
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
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
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…
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