Asterisk cmd Milliwatt
Posted: June 5, 2005 Synopsis: Generate a Constant 1004 Hz tone at 0dbm ( mu-law ) Description: Milliwatt() Generate a Constant 1004 Hz tone at 0dbm (mu-law) Earlier versions of Asterisk generated a 1000 Hz Tone. If the old behaviour is desired…
Read More »Zultys phones
Posted: June 5, 2005 Company web site: https://www.zultys.com/ VOIP phones, media gateways and softphones. Their LIPZ4 Linux softphone is free. How to configure Asterisk for LIPZ4 1.3.11 The cheap (?) ZIP 2 phone ($100) comes without any display, but all other typical…
Read More »Asterisk Slimming
Posted: June 5, 2005 Slimming Asterisk Footprint by Eliminating Unneeded Dynamic Modules The stock installation of asterisk results in many dynamically loaded modules, providing the full suite of channel types, codecs, file formats, application commands, and accounting/database interfaces. Eliminating unneeded dynamic modules…
Read More »Teamspeak
Posted: June 4, 2005 TeamSpeak http://www.goteamspeak.com TeamSpeak is an application which allows its users to talk to each other over the internet and basically was designed to run in the background of online games. The voice quality and hereby the bandwidth usage…
Read More »cnum.info
Posted: June 4, 2005 cnum.info – Meta-ENUM and LCR CNUM.INFO – Meta ENUM and LCR cnum.info is a domain used for ENUM-stylish-queries to provide realtime SIP routing information based on PBX admin data for low coast communication. How it works cnum.info’s nameserver…
Read More »LCR tool for i4l
Posted: June 4, 2005 Least Cost Routing Tools for i4l, asterisk and others Asterisk addon rate-engine From Voop (previously Trollphone) comes the Rate-engine to allow smart LCR routing. For more info, see link above Meta-ENUM lookup for combined Call and LCR routing…
Read More »DrayTek VoIP Routers
Posted: June 4, 2005 Wireless Routers with built-in VoIP DrayTek Vigor2930Vn Dual-wan with fail-over/load-balance SIP, RTP/RTCP protocols compliance 2 FXS ports supporting up to 12 SIP registrars Integrated IEEE802.11g/b/n WLAN AP QoS: Guaranteed bandwidth for VoIP Built-in SPI Firewall and up to…
Read More »Asterisk timer ztdummy
Posted: June 4, 2005 How to compile ztdummy ztdummy is a Linux kernel module that will provide your Asterisk with a Zaptel timer even if you don’t have any Digium hardware installed in your Linux server. For Linux kernel version 2.4 the…
Read More »Asterisk password files
Posted: June 3, 2005 Asterisk passwords are scattered all over the place. In the channels configuration files for clients (phones) In the voicemail system for voicemailboxes In a separate file for the Asterisk cmd Authenticate command In a file for the DISA…
Read More »Asterisk for Medical Offices
Posted: June 3, 2005 Asterisk for Medical Offices Asterisk is an enterprise-capable PBX. It runs with both digital and analog telephony connections and hardware. This document is a summary of a system that would provide a full range of computer telephony services…
Read More »Asterisk cmd GetGroupMatchCount
Posted: June 3, 2005 Synopsis GetGroupMatchCount(groupmatch[@category]) Version New with Asterisk 1.1 development branch. This command is deprecated in Asterisk v1.2. Instead use GROUP_MATCH_COUNT(groupmatch[@category]) Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)}) Description Calculates the group count for all groups that match the specified pattern. Uses standard regular expression matching…
Read More »Leadtek
Posted: June 3, 2005 Leadtek is an OEM manufacturer for video phones and SIP ATAs. http://www.leadtek.com/ Also See This Leadtek Page Support: Leadtek Support Forum Their video phone products are sold by: ProVu Communications Packet8 etc. Their SIP ATA models are: Leadtek…
Read More »Asterisk indications Croatia
Posted: June 2, 2005 Asterisk indications for Croatia Insert this into indications.conf and set the Asterisk cmd SetLanguage to hr. [hr] description = Croatia (Hrvatska) ringcadence = 1000,4000 ; Internal Dialtone (used on PBX) ;dial = 425 ; External Dialtone (used on…
Read More »Asterisk tips simple wake-up call
Posted: June 2, 2005 Yet another wake-up call implementation I put together this perl/AGI script instead of installing one of the others for these reasons: No cron job required Works the same way as the PSTN “built-in” wake-up service has “always” worked…
Read More »Asterisk bounty Call Pickup with CID info. – SIP
Posted: June 2, 2005 Call Pickup with CID info. (SIP) When a call is transferred from a ringing SIP-Phone to an other SIP-Phone using the *8 feature no call info is displayed. This is offcourse logical because the set is dialing out…
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