DUNDi peers tier3 us fl
Posted: October 22, 2004 Florida Tier 3 Peers Please use the peer listing format to list your information. Use the following links to locate a Tier 3 peer: Pro-Medical, Inc. BANDWIDTH: 1.54 mb/s PUBKEY: COUNTRY: UNITED STATES STATEPROV: FL AREACODE: 352 CONTACT:…
Read More »Asterisk cmd DUNDiLookup
Posted: October 22, 2004 Note from Asterisk 1.2: Jul 27 14:19:08 WARNING: pbx_dundi.c:3872 dundi_lookup_exec: This application has been deprecated in favor of the DUNDILOOKUP dialplan function Synopsis: Look up a number with DUNDi Description: DUNDiLookup(number[|context[|options]]) Looks up a given number in the…
Read More »IAXComm
Posted: October 22, 2004 IAXComm IAX softphone for Windows, Linux & Mac OS X Features: Register with multiple servers (ie enterprise server and iaxtel). Multiple call appearances. User selectable audio devices. User defined ringtones. Autoanswer intercom calls (with password protection). Registration with…
Read More »Asterisk cmd UserEvent
Posted: October 20, 2004 UserEvent Synopsis Send an arbitrary event to the manager interface Description UserEvent(eventname[|body]) Sends an arbitrary event to the manager interface, with an optional body representing additional arguments. To split body text to several lines | character could be…
Read More »Asterisk sip qualify
Posted: October 19, 2004 SIP.conf: device configuration – qualify Syntax: qualify=xxx|no|yes where XXX is the number of milliseconds used. If yes the default timeout is used 2 seconds. If you turn on qualify in the configuration of a SIP device in Asterisk config…
Read More »H.323 Forum
Posted: October 19, 2004 The H.323 protocol suite is an ITU standard addressing multimedia communication. Today, H.323 enjoys the majority share of Voice Over IP market and a growing share of Video Over IP market. Carrying traffic measured in billions of minutes…
Read More »TRIP
Posted: October 19, 2004 From the RFC 3219 : This document presents the Telephony Routing over IP (TRIP). TRIP is a policy driven inter-administrative domain protocol for advertising the reachability of telephony destinations between location servers, and for advertising attributes of the…
Read More »Megaco
Posted: October 19, 2004 Megaco (Media Gateway Control) is a control and signalling protocol for VOIP connections. Megaco was jointly developed by the IETF and the ITU. H.248 is the ITU name for Megaco. Megaco was developed from MGCP and as a…
Read More »Asterisk cmd ForkCDR
Posted: October 19, 2004 ForkCDR Synopsis Forks the Call Data Record ForkCDR() Causes the Call Data Record to fork an additional cdr record starting from the time of the fork call. This new cdr record will be linked to end of the…
Read More »Asterisk cmd ZapScan
Posted: October 19, 2004 ZapScan Synopsis Scan Zap channels to monitor calls Description ZapScan allows a call center manager to monitor Zap channels in a convenient way. Use ‘#’ to select the next channel and use ‘*’ to exit Asterisk 1.6 :…
Read More »Asterisk cmd ZapBarge
Posted: October 19, 2004 Synopsis Listen to a Zap channel call Description ZapBarge(channel) Lets you listens to the conversation on a specified Asterisk ZAP channels, or prompts if one is not specified. You can hear them, but they can’t hear you. No…
Read More »Asterisk cmd ResetCDR
Posted: October 19, 2004 Synopsis: Resets the Call Data Record Description: ResetCDR([options]) Causes the Call Data Record to be reset, optionally storing the current CDR before zeroing it out (if ‘w’ option is specifed). A CDR record will be stored for any…
Read More »Asterisk cmd NoCDR
Posted: October 19, 2004 Synopsis: Make sure asterisk doesn’t save CDR for a certain call Description: NoCDR() Makes sure there won’t be any CDR written for a certain call See also CDR Asterisk billing Asterisk cmd ResetCDR Asterisk cmd ForkCDR Asterisk |…
Read More »PlaySip
Posted: October 19, 2004 Description “playSIP” is a command-line program that can be used to call a SIP session (i.e., specified by a URL beginning with “sip:”), and (optionally) record the incoming media stream into a file. Details “playSIP” is run the…
Read More »Asterisk howto dial plan
Posted: October 18, 2004 How to build a dial plan in Asterisk The dial plan is built in extensions.conf and included files. Basic terms A dial plan consists of a number of extensions. Each extensions consists of a number of priorities. Extensions…
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