
Author: VoIP Info
Asterisk cmd DynExtenDB
What is DynExtenDB? DynExtenDB is a simple solution to manage a huge amount of extensions which change quite often. DynExtenDB is something like your old extensions.conf, but it changes the way extensions are used. Your extensions will (additionally…
Read More »Asterisk cmd AgentMonitorOutgoing
Syntax: AgentMonitorOutgoing([options]) Tries to figure out the id of the agent who is placing outgoing call based on comparision of the callerid of the current interface and the global variable placed by the Asterisk cmd AgentCallbackLogin application. That’s…
Read More »World Phone
The World Phone works just like the telephones you currently use. Dial tones, busy signals, and call waiting notification are all supported. . Clear communications on broadband AND dial up connections, anywhere in the world.. Connection is through…
Read More »SIP method message
From the Abstract of RFC 3428: Instant Messaging (IM) refers to the transfer of messages between users in near real-time. These messages are usually, but not required to be, short. IMs are often used in a conversational mode,…
Read More »PBX Skill Based Routing
Skills Based Routing Definition The following is the Wikipedia definition for Skills Based Routing: “Skills-Based routing (SBR) is a call assignment strategy used in call centers to assign incoming calls to the most suitable agent, rather than simply…
Read More »Asterisk 0.7.0
Changes in 0.7.0 since 0.5.0 VoiceMail and VoiceMailMain were discontinued in favour of VoiceMail2 and VoiceMailMain2IAX was discontinued in favour of IAX2; to completely remove IAX1 support from Asterisk it is suggested to include “noload => chan_iax.so” in…
Read More »RFC 3311
The Session Initiation Protocol (SIP) UPDATE Method Abstract This specification defines the new UPDATE method for the SessionInitiation Protocol (SIP). UPDATE allows a client to updateparameters of a session (such as the set of media streams and theircodecs)…
Read More »Asterisk OpenBSD patch
Asterisk 0.7.1 on OpenBSD 3.4 For use asterisk on OpenBSD i’ve create a patch, because a lot of compilation error append. Use ‘gmake’ not ‘make’ Like lot of other linux software you must use gmake (if you don’t…
Read More »Asterisk OpenPSD patch
Asterisk 0.7.1 on OpenBSD 3.4 For use asterisk on OpenBSD i’ve create a patch, because a lot of compilation error append. Use ‘gmake’ not ‘make’ Like lot of other linux software you must use gmake (if you don’t…
Read More »IAX Phone
This product appears to have vanished, there is no longer any mention of it on the web site. IAX Phone is an IAX2 soft phone. It is currently available as a free demo. Two GA versions should be…
Read More »Telebau Phones
H.323 phones IP Tel 100: H.323 IP Tel 102: H.323 with 1 S0 gateway (for ISDN BRI) Gateways and adaptors IP Gate 104: 2 x ISDN BRI S0 (max. 6) to 1 x LAN gateway (H.323) IP Gate…
Read More »SER example pstn
SER managing a telephony gateway example: ser configured as PSTN gateway guard; PSTN gateway is located at 192.168.0.10 $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $ —————— module loading ———————————- loadmodule “modules/sl/sl.so” loadmodule “modules/tm/tm.so” loadmodule “modules/acc/acc.so” loadmodule “modules/rr/rr.so”…
Read More »SER module enum
Enum module implements enum_query function that makes an ENUM query based on the user part of the current request URI. The function assumes that the user part consists of an international phone number of the form +decimal-digits, where…
Read More »VoIP User Group San Antonio Texas USA
San Antonio VoIP User Group We are in the process of forming the Austin VUG. The plan is to meet once a month for about 2 hours. The proposed meeting format is: Breaking news, 15 minutes Presentation by…
Read More »VoIP User Group Houston Texas USA
Houston VoIP User Group We are in the process of forming the Austin VUG. The plan is to meet once a month for about 2 hours. The proposed meeting format is: Breaking news, 15 minutes Presentation by a…
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