Hmm, well, the Dial() command in 1.2 has the power to do call screening: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial#Dialmacros
And it looks like features.conf _might_ let you interrupt a call in progress using the [applicationmap] section, though I haven't gotten that to work yet: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf Don't ask me why there's only a ZapBarge, and no generic Barge... If you go for the conferencing solution, I suggest you build cmd_conference, as it's much lighter weight, and doesn't have all the timing issues meetme does. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference There's an example there on how to send random announcements into the conference using AMI on that page too. re, spd On Wed, 5 Sep 2007, Mike C. Fletcher wrote: > [EMAIL PROTECTED] wrote: > > I think you can do something like that with chan_local > > > If I'm reading correctly, chan_local will immediately drop out when the > SIP channel is connected to it. That is, you have caller A "dial" the > local context, it starts processing a dialplan, but the moment it > completes a Dial() out to the user, the two calls are linked and cease > processing in their dialplan (i.e. on success they cease dialplan > processing, but if the dial fails they continue in the dialplan). I'm > trying to intercept the second call *after* the dial-out completes, that > is, the callee would get connected to an IVR that would allow for opting > out of all future calls, *then* the link would go through. > > For background, this is what I'm looking at: > > * caller one gets his call, quick "hello" and record your name, then > starts ringing caller 2 (continue ringing until bridge) > * caller two gets her call, IVR says "hi, you've got a call from" > name, if you'd like to accept, press X, if you'd like to reject, press > Y, if you'd like to ban this user from calling you again, press Z, if > you'd like to prevent all future calls from all users, press A, if they > choose X, they get linked with the caller (whose ringing ears now stop > ringing). > * after a given period, the two callers hear a "sorry, your time is > running out" and then a few seconds later a cheery "goodbye" to both > * preferably, at any point during the call, caller two (the callee) > could invoke the options to bar/ban caller one > > At the moment I don't see how to do do that with chan_local... but then > I've been dense before :) . At the moment the only approach seems to be > using MeetMe (or whatever) to bridge the two connections once they've > gone through their respective IVRs to allow for the escapes to menus and > playing the sounds (given the seemingly broken L option on my machine > (still no luck there)). > > You would think this could be a matter of: > * dial each user into a context > * provide IVR introduction > * monitor the channels for IVRs, on IVR un-bridge the channels if > bridged already and have each continue in their previous contexts with > the IVR selection available > * bridge the channels > * play sound into channel at X time > * un-bridge after X period with '' as the IVR selection > > Oh well, that's life. > > Have fun, > Mike > > On Wed, 5 Sep 2007, Mike C. Fletcher wrote: > ... > >> In another "heh, > >> wouldn't this be an obvious first thing to expose" moment, it looks like > >> the ami command "Bridge" is only available in Asterisk 1.6+. > >> Double-sigh. I gather there must be a reasonable way to take two > >> outgoing legs and join them with a short dialplan IVR interposed before > >> you put either caller into the bridge... but I don't see it yet short of > >> MeetMe or the like. > >> > ... > > -- > ________________________________________________ > Mike C. Fletcher > Designer, VR Plumber, Coder > http://www.vrplumber.com > http://blog.vrplumber.com > > --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
