login | register
Wed 01 of Aug, 2007 [09:23 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Last Changes
Shoutbox
  • 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: 0.42s
  • Memory usage: 2.19MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 3.06

Asterisk cmd FollowMe

FollowMe

Synopsis

Find-Me/Follow-Me application
Introduced with Asterisk 1.4

Description

FollowMe(followmeid|options):

This application performs Find-Me/Follow-Me functionality for the caller as defined in the profile matching the <followmeid> parameter in followme.conf. If the specified <followmeid> profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority.
Forked (=simultaneous) dialing of multiple numbers in the same step is supported with this application if you'd like to dial multiple numbers in the same followme step.

 Options:
   s    - Playback the incoming status message prior to starting the follow-me step(s)
   a    - Record the caller's name so it can be announced to the callee on each step
   n    - Playback the unreachable status message if we've run out of steps to reach the
          or the callee has elected not to be reachable.

Example

...work in progress...

See also


Go back to Asterisk

Created by peletiah, Last modification by JustRumours on Tue 27 of Mar, 2007 [21:22 UTC]

Comments

Comments Filter

by shido6 on Wednesday 31 of January, 2007 [23:35:29 UTC]
Doesnt seem to work for me, shouldnt it be using "Background" instead of "Playback" ?

Where is it listening for my input?

-sx

Request for Feature

by Mark Rounds on Sunday 29 of October, 2006 [04:12:57 UTC]
I don't suppose that we could get a dynamic access to the follow me's like the switch statement in extensions.conf. This would allow for cleaner web interfaces to be built for adding and changing the follow me for users. Just a thought.

Request for Feature

by Mark Rounds on Sunday 29 of October, 2006 [04:12:35 UTC]
I don't suppose that we could get a dynamic access to the follow me's like the switch statement in extensions.conf. This would allow for cleaner web interfaces to be built for adding and changing the follow me for users. Just a thought.

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver