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Wed 01 of Aug, 2007 [09:27 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
  • Execution time: 0.32s
  • Memory usage: 2.17MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 2.34

Telebau Phones

Telebau

http://213.160.66.220/voip/index.html

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 208: 4 x S0 (max. 6) to 2 x LAN gateway (H.323)
  • IP Gate 312: 6 x S0 to 3 x LAN gateway (H.323)
  • IP TA-102-4: 4 x analog, 1 x S0, 3 x door to 1 x LAN

Remarks

taken from (in German): Heise forum
to-pse (22. Januar 2004 11:15)

Wir haben diese Geräte (und ähnliche mit dem gleichen Protokollstack)
hier im Haus im Test gegen unseren eigenen H.323-Stack.

Davon abgesehen, dass kein standardisiertes H.225 RAS Protokoll
zur Rufnummernauflösung verwendet wird, sind die Telefone eigentlich
ganz tauglich. Die nicht standard-konforme Rufnummernauflösung
verhindert aber, dass andere H.323 Produkte mit diesen Geräten
über einen Gatekeeper Verbindungen etablieren. Direkter Anruf
über IP-Adresse geht aber.

Allerdings gab es mit älteren Softwareständen auch in anderen H.225
und H.245 Bereichen ein paar Probleme. Diese sind allerdings in-
zwischen vom Hersteller des Stacks behoben... also auf Firmware-
Updates achten!


Go back to VoIP phones

Created by JustRumours, Last modification by JustRumours on Thu 22 of Jan, 2004 [11:59 UTC]

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