Followme can be configured to accept a keypress to accept or decline the call. Put something like this in followme.conf:
[general] featuredigittimeout=>5000 takecall=>1 declinecall=>2 call_from_prompt=>followme/call-from norecording_prompt=>followme/no-recording options_prompt=>followme/options pls_hold_prompt=>followme/pls-hold-while-try status_prompt=>followme/status sorry_prompt=>followme/sorry On Thu, Apr 28, 2016 at 8:46 AM, A J Stiles <[email protected]> wrote: > On Thursday 28 Apr 2016, Robin Kipp wrote: > > > Hi all, > > > > > > sorry if the subject is a bit confusing, but I just couldn’t think of a > > > good way of better describing the situation… > > > > > > Basically, I travel a lot and have several SIM cards for my phone from > > > local carriers. What I’d like to do now is to setup Asterisk, so that > > > people who want to reach me just have to dial one number which forwards > > > the call to all my cellphone numbers in turn. I’m still pretty new to > > > Asterisk, so I’m unsure which method would be most suitable for this > > > scenario. > > > > > > Theoretically, I could use the dial function to call one number, then > wait > > > a few seconds and then dial another number. In practice, this won’t work > > > because as soon as a call is answered by the mobile carrier’s voicemail > > > the caller would be connected to that, no other numbers would be called. > > > So here’s my question: how can I possibly avoid this situation? Is there > a > > > way for Asterisk to detect such situations and distinguish them from me > > > actually trying to answer the call when the correct number is called? Not > > > sure if this is technically possible, but figured I’d ask just in case > > > there is any sort of solution. I’m aware that it would be best to simply > > > use SIP and a SIP client on my phone in order to take the call, but due > to > > > most carriers blocking SIP traffic on their mobile data networks this > > > wouldn’t work as soon as I’m not connected to any WiFi. So, in case > > > there’s any solution to this problem I’d greatly appreciate if you could > > > share that with me! Many thanks and best wishes, > > > Robin > > > > There is no reliable way to distinguish whether a phone was answered by a > human being or a machine. > > > > If you can't just disable voicemail on all your SIMs then you will need to > find out how long each carrier will wait before diverting to voicemail, and > then make sure the timeouts in your Dial() statements are short enough not > to trigger the carrier's voicemail. Then use Asterisk's VoiceMail() > application to record any message your caller might leave. > > > > You can just chain a whole bunch of Dial() statements one after another > within an extension because once the first one has been answered, execution > will proceed to the "h" extension. > > > > If you install ConnectBot on your mobile phone, you should be able to > login to your Asterisk server each time you swap in a new SIM card, and > edit your dialplan so the phone you are using today gets tried first. This > could even be automated, but talk of such may not be appropriate for a > non-commercial list. > > > > Out of politeness to the caller, play them a recorded announcement before > your bank of Dial() statements, so they know to wait while your Asterisk > box searches for you. > > > > Finally, remember: You will be tying up two channels -- and therefore > maybe two DAHDI spans, depending how the calls are coming into and out of > your Asterisk box -- with this. > > > > -- > > AJS > > > > Note: Originating address only accepts e-mail from list! If replying > off-list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
