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

Discussion in 'VOIP' started by Vincent Delporte, Jun 22, 2006.

  1. Hello

    (sorry, I know this ng is aimed at VoIP, and what I'm trying to
    achieve first is closer to telecom than VoIP, but I'm getting nowhere,
    so figured maybe someone would know the answer. Thx)

    Since I'm stuck, I went back to reading several PDFs on Asterisk, and
    I'm beginning to wonder if it's at all possible to have Asterisk ring
    a phone number without first answering the call and asking the user to
    type an extension.

    I have two FXO cards: When a call comes into the first card, I want
    Asterisk to simply dial out a number through the second card without
    going off hook.

    Anybody knows if I'm just wasting my time with Asterisk to do this,
    and should look at another solution? All the exemples I see of dial
    plans include extensions, ie. callers are expected to go through some
    kind of voice menu and choose an extension for the magic to happen.

    Thank you.
     
    Vincent Delporte, Jun 22, 2006
    #1
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  2. Vincent Delporte

    airdog Guest

    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)
     
    airdog, Jun 22, 2006
    #2
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  3. 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.
     
    Vincent Delporte, Jun 23, 2006
    #3
  4. Vincent Delporte

    Heimo Hetl Guest

    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
     
    Heimo Hetl, Jun 23, 2006
    #4
  5. 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.
     
    Vincent Delporte, Jun 23, 2006
    #5
  6. [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 [email protected] help forum:

    /*
    What you are looking for is a feature called DISA.
    Read the following:
    http://www.voip-info.org/wiki-Asterisk+cmd+DISA
    http://nerdvittles.com/index.php?p=73

    For that matter, read EVERYTHING here (Ward has been VERY explicit):
    http://nerdvittles.com/index.php?p=130
    */

    Also see https://sourceforge.net/forum/message.php?msg_id=3737732
     
    William P.N. Smith, Jun 24, 2006
    #6
  7. 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;
    http://www.rpath.org/rbuilder/project/asterisk/) 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

    http://codecomplete.free.fr/asterisk/

    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!
     
    Vincent Delporte, Jun 24, 2006
    #7
  8. Vincent Delporte

    Bill Kearney Guest

    In his case, Asterisk is set up so that, if his office phone doesn't
    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?
     
    Bill Kearney, Jun 27, 2006
    #8
  9. 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.
     
    Vincent Delporte, Jun 27, 2006
    #9
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