Author: VoIP Info
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…
Read More »sipX
Posted: June 2, 2005 sipX – The SIP PBX for Linux Original project: sipXecs Original Website: http://www.sipfoundry.org This project has also been forked and continues with the original developers under the sipXcom name. New project: http://www.sipxcom.org Wiki: http://wiki.sipxcom.org User Forums: https://groups.google.com/forum/#!forum/sipxcom-users See…
Read More »Asterisk cmd BristuffPickUpChan
Posted: June 2, 2005 PickUpChan Synopsis Channel independent call pickup. Description PickupChan(Technology/resource[&Technology2/resource2…]) Tries to pickup the first ringing channel in the parameter list. Related applications Asterisk cmd Pickup: As included with standard Asterisk Bristuff PickUp: Mostly channel independent; part of Bristuff patch…
Read More »Asterisk cmd BristuffPickDown
Posted: June 2, 2005 PickDown Synopsis Channel independent call pickdown. Description PickDown([group]) Tries to hangup the first ringing channel with callgroup == group. If called without the group argument, the pickupgroup of the channel will be used. Related applications PickUp: Mostly channel…
Read More »Lucent TNT
Posted: June 2, 2005 Lucent Max TNT is used in large deployments as it supports multiple T1/E1 interfaces. Takes 10U worth of rackspace! It Can talk to Asterisk via both H.323 & SIP Here is the TNT config to talk SIP: Using…
Read More »Asterisk SIP not-proxy
Posted: June 1, 2005 Why is Asterisk not a SIP Proxy? Asterisk is *not* a SIP proxy. A SIP proxy handles call control on behalf of other user agents (UA) and usually does not maintain state during a call and therefore is…
Read More »Asterisk How to connect to Nikotel
Posted: June 1, 2005 From the Asterisk Mailing list: register=username:[email protected]/#your_extension# [nikotel] type=friend host=calamar0.nikotel.com username=##username## secret=******* fromuser=##username## fromdomain=nikotel.com qualify=yes context=nikotel-inbound insecure=very ; required if you want to accept incoming Nikotel calls from host promiscredir=yes ; required to handle the Nikotel 302 redirect message…
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Phone System 