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

 
 
Vincent Delporte
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      06-22-2006
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.
 
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airdog
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      06-22-2006
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)

On Thu, 22 Jun 2006 21:51:46 +0200, Vincent Delporte wrote:

> 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.


 
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Vincent Delporte
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      06-23-2006
On Thu, 22 Jun 2006 22:41:45 GMT, airdog <(E-Mail Removed)> wrote:
>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.
 
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Heimo Hetl
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      06-23-2006
airdog <(E-Mail Removed)> wrote:

> 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)


You can even skip the Answer().

> On Thu, 22 Jun 2006 21:51:46 +0200, Vincent Delporte wrote:


> > 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.


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

> > 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.


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

--
You never ask questions when God's on your side.
 
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Vincent Delporte
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      06-23-2006
On Fri, 23 Jun 2006 03:01:28 +0200, http://www.velocityreviews.com/forums/(E-Mail Removed) (Heimo Hetl)
wrote:
>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.


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.
 
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William P.N. Smith
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      06-24-2006
Vincent Delporte <(E-Mail Removed)> wrote:
>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.


[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:
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/messag...msg_id=3737732

 
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Vincent Delporte
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      06-24-2006
On Fri, 23 Jun 2006 21:09:53 -0400, William P.N. Smith
<(E-Mail Removed)> wrote:
>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.


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 ------------------

>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...


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!
 
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Bill Kearney
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      06-27-2006
> 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.


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?

 
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Vincent Delporte
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      06-27-2006
On Tue, 27 Jun 2006 10:44:50 -0400, "Bill Kearney"
<(E-Mail Removed)> wrote:
>Which voicemail? The one in from his cell phone provider?


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.

>Does the caller ID info from a call forwarded this way indicate the cell phone number
>or the original caller?


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.
 
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