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Wed 01 of Aug, 2007 [09:19 UTC]

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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
Server Stats
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  • Database queries: 30
  • GZIP: Disabled
  • Server load: 3.11

BSD

The FreeBSD Project

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Other FreeBSD pages on voip-info.org:

Asterisk FreeBSD info specific to running Asterisk on FreeBSD

VoIP technologies available on FreeBSD

  • Asterisk - the open source PBX and multi-protocol VoIP server
  • OpenPBX - an open source PBX with support for OH323 PSTNGW
  • SIP Express Router (SER) - the open source SIP proxy server
  • Aqua Gatekeeper - Aqua H323 Gatekeeper and proxy
  • Skype - P2P VoIP software
  • KPhone - A voice over internet phone
  • Vomit - Voice over misconfigured internet telephones
  • cphone - H323 Video Conferencing Program which uses QT
  • GnomeMeeting - GNOME H323 Video Conferencing program, similar to NetMeeting
  • Open H.323 - Opensource H.323 toolkit
  • OpenMCU - OpenMCU hosts a conference call for H323 Video Conferencing users


please add more items.


The OpenBSD Project

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VoIP technologies available on OpenBSD

please add more items.

OpenDarwin

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VoIP technologies available on OpenDarwin

  • Asterisk - the open source PBX and multi-protocol VoIP server
please add more items

NetBSD

I guess someone decided to nuke the link to the best BSD ;-)
Anyway, here's the wiki page.
Created by benjk, Last modification by Torben Brosten on Mon 14 of May, 2007 [20:36 UTC]

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