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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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OpenPBX.org Glossary

Glossary of terms used on OpenPBX.org wiki pages


AGI

  • Asterisk Gateway Interface, an interface to run external scripts within the Asterisk PBX' dialplan.

API

  • Application Programming Interface

BRI

  • Basic Rate ISDN

CLI

  • Command Line Interface

DSP


GPL


H.323


IAX2

  • Inter Asterisk eXchange, a VoIP protocol initially designed for peering Asterisk servers

IP


IRC


ISDN


IVR


Jingle

  • Jingle is a VoIP extension to the XMPP/Jabber instant messaging protocol

LTP

  • Lightweight Telephony Protocol - a signaling and media protocol for VoIP

MFC/R2

  • Multi-Frequency Code signaling Release 2, a channel associated signaling protocol

MGCP


NAT

  • Network Address Translation - a scheme to route traffic between private non-routable IP addresses on a LAN and public IP address on the internet

OGI

  • OpenPBX Gateway Interface, same as AGI, but renamed for use with OpenPBX.org

PBX

  • Private Branch eXchange

POSIX


POTS

  • Plain Old Telephone System, usually referring to analog telephone lines

PRI

  • Primary Rate ISDN

PSTN

  • Public Switched Telephone Network

RC

  • Release Candidate

RTP


SIP


SCCP

  • Skinny Client Control Protocol, a proprietary signaling protocol for VoIP by Cisco

STUN


SVN

  • short name for the Subversion revision control system by CollabNet

T.38

  • a fax over IP standard

UDP


VoIP

  • Voice over Internet Protocol - a generic term for protocols used to carry telephone calls over IP.

XMPP


Zaptel

  • Zapata Telephony interface, initially for use with telephony hardware designed by the Zapata project


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Created by STS, Last modification by STS on Fri 08 of Dec, 2006 [06:15 UTC]

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