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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: 0.24s
  • Memory usage: 2.18MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 3.75

Asterisk vertical service activation codes

for analog phones (ZAP channel)


*0 Flash external trunk on bridged channel.
*60 Blacklist last caller (if callerid was present) (after 13 Sep 2004)
*67 Disable Caller ID for next outgoing call (per call blocking).
*69 Call return. Dials number of last caller if caller ID was present.
*70 Disable call waiting for the next call or until hangup.
*72 Enable call forwarding.
*73 Cancel call forwarding.
*77 Enable Anonymous Call Rejection
*78 Enable do not disturb.
*79 Disable do not disturb.
*80 Blacklist the caller who called previously (If Caller ID was present) (before 13 Sep 2004)
*82 Enable caller ID on a line with per-line blocking.
*87 Disable Anonymous Call Rejection

Note: These codes are the US numbering sequences. If you like European style codes -for now- you'll need to implement them in the dialplan. See some of the examples in the Dial plan solutions at Asterisk tips and tricks.

See also


Created by hwstar, Last modification by fmandarino on Tue 02 of Nov, 2004 [16:42 UTC]

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