[Asterisk] Ringing a remote phone WITHOUT typing an extension first?

You just need to use the Answer command in Asterisk, and then dial the extension you want it forwarded to.

So it would go something like this:

exten => s,1,Answer() exten => s,2,Dial(chanXX/extenXX)

Thanks for the input... but like I said, I don't want Asterisk to go off-hook needlessly: In case no one answers the call in either location, the caller will end up paying for a call that never made it through. If Asterisk really can't handle a call without going off-hook, I'll have to take the call, play some kind of ring tone to the caller while Asterisk rings the phones in both locations, and play a message if no one picked it up.

Considering how rich Asterisk is, I'm surprised it can't do this, though.

Thanks for the help.

You can even skip the Answer().

Asterisk dials if you call Dial(). No need to Answer() first. And no need for any user interaction, either.

You misunderstood Asterisk's concept of extensions. An extension is simply a combination of a channel and an address. It is a definition of who to call and by which means.

cheers Heimo

I think I did understand, but 1) all the examples I see deal with a voice menu and expect the caller to type an extension, and 2) I already tried the example you gave: Asterisk goes off-hook, and remains silent instead of using FXO #2 to dial a remote location.

Does someone have an actual configuration of what I'd like to do? Again, if possible, I'd like to avoid having Asterisk go off-hook and force the caller to pay needlessly for a call that no one answered.

Thank you.

[This is based on my very limited knowledge of Asterisk @ Home V2.8]

You can do this with an extension (just point incoming calls at a ring group, and the incoming call won't be picked up till one of the extensions in the ring group picks up [or it rings too many times and goes to IVR or voicemail, but you could set your ring timeout to four hours or something]).

However, I don't think you can do this with outgoing POTS lines, as I don't think you can tell when someone picks up on the other end. 'Supervision' on a POTS line seems to happen as soon as the call is dialed.

On the other tentacle, I think you _can_ do this with a VOIP provider that provides supervision when the calling party picks up, and then add your external phone number to the ring group. You may need to try VOIP providers to find one that'll provide the proper supervision...

My network is down as I'm typing this, so I'm not certain it applies, but I found the following in the A@H help forum:

/* What you are looking for is a feature called DISA. Read the following:

For that matter, read EVERYTHING here (Ward has been VERY explicit): */

Also see

Thanks a lot for the explanation :-)

The problem is that I'm getting conflicting feedback: Someone told me elsewhere that he did set up his Asterisk server to do just what I want.

In his case, Asterisk is set up so that, if his office phone doesn't answer within X rings, Asterisk will then dial() his cellphone, and if it still doesn't get an answer, ends up in his voicemail.

That's exactly what I want, but all I'm getting so far running rPath Linux PoundKey (ready-to-use Asterisk disto using Asterisk 1.2.5;

is Asterisk going off-hook, and silence (actually, a mix of silence and crap that sounds like static).

FWIW, you'll have the configuration files and a couple of log files at

Here's what the console says when I call in:

--------- LOG ------------------ Connected to Asterisk 1.2.5 currently running on localhost (pid =

1790) Verbosity is at least 3 -- Starting simple switch on 'Zap/1-1' Jun 23 18:39:42 NOTICE[2445]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Jun 23 18:39:44 NOTICE[2445]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)... Jun 23 18:39:47 NOTICE[2445]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Dial("Zap/1-1", "Zap/2/01XXXXXXXX|30|r") in new stack -- Called 2/01XXXXXXXX -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 *CLI> stop now

Beginning asterisk shutdown.... -- Hungup 'Zap/2-1' == Spawn extension (cherbourg, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). )

--------- LOG ------------------

If Asterisk really can't handle this scenario (maybe no one else used two FXO clones to just route calls like this before?), I'll look into using SIP phones instead.

Thx for the help!

Which voicemail? The one in from his cell phone provider?

I'd wonder about the possibility of handing the call off to the cell phone and then letting the cell phone had it back to the incoming line. Does the caller ID info from a call forwarded this way indicate the cell phone number or the original caller?

Whichever, I don't mind. But we're already three people who unsucessfuly tried to do what I described in the original post. Considering the number of Asterisk servers in use today, I'm very surprised that we're the only ones to have ever needed to have Asterisk ring a remote phone and bridge the call through two FXO cards.

I can't tell yet, as Asterisk is stuck: It goes off-hook (even without any Answer() in the context), and remains silent. More information tomorrow if I can spend some time on it.