asterisk users - Feb 2006 - SV: SV: delaying "answer" for a number of ringsor anamount of time

>From what I understand it means that the *hardware* in your computer
*acknowledges* the call as soon as it is recieved and then sends it to asterisk
dialplan for processing.
You would essentially need to put the delay before the call ever reaches
asterisk. So this problem isn't asterisk related... if I've understood
your question and the answer I found correctly.

Regards,
Jan

-----Ursprungligt meddelande-----
Fr?n: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] F?r Brian J. Murrell
Skickat: den 2 februari 2006 22:37
Till: asterisk-users@lists.digium.com
?mne: Re: SV: [Asterisk-Users] delaying "answer" for a number of
ringsor anamount of time

On Thu, 2006-02-02 at 22:08 +0100, jan.sarin@securia.se
wrote:
> http://lists.digium.com/pipermail/asterisk-users/2005-September/125146
> .html
OK.  The hardware is a wildcard though.  How does that answer apply?
Isn't it asterisk itself that is picking that call up?  Can't it delay
the pick up?  Maybe I am just misunderstanding your reference.

b.

--
My other computer is your Microsoft Windows server.

Brian J. Murrell

On Fri, 2006-02-03 at 08:49 +0100, jan.sarin@securia.se
wrote:
> >From what I understand it means that the *hardware* in your computer
*acknowledges* the call as soon as it is recieved and then sends it to asterisk
dialplan for processing.
Hrm.  Yes, that is what I got from it.  But in my case the hardware is
an internal, PCI (Zap) Wildcard.  I am pressuming that since I can use
functions like "Wait()", then "Answer()" in dialplan to
actually delay
answering (for the Wait() time) that Asterisk actually "acknowledges"
the call.
> You would essentially need to put the delay before the call ever reaches
asterisk. So this problem isn't asterisk related... if I've understood
your question and the answer I found correctly.
Hrm.  Yeah.  Perhaps.  I guess perhaps Asterisk isn't currently able to
handle deciding if the call has been hung up before it even picks it up
(i.e. no more rings).  Maybe a peek at the source to Asterisk and the
zaptel drivers might tell me more.

I just find it strange that I am the first person to want this feature.
Indeed tomshardware.com has an article describing how to make an
answering machine out of Asterisk by doing exactly what I tried.  In my
experiments though, it just don't work the way they describe it.

Thanx,
b.

-- 
My other computer is your Microsoft Windows server.

Brian J. Murrell
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20060203/e4b89721/attachment.pgp

Hi,

We are ordering a bank of numbers from our provider BT.  We will have an 
ISDN30 with 8 channels enabled.  Is it possible to do this? Is this known 
as DDI?  Can anyone give tips on how to configure the Asterisk server so 
that users are available on the extensions. 

Hope this explains this better ...

01925 838381            Switchboard
01925 838382            User 1
01925 838383            User 2
01925 838384            User 3
01925 838385            User 4
01925 838386            User 5
01925 838387            User 6
01925 838388            User 7
01925 838389            User 8
01925 838390            User 9
01925 838391            User 10
01925 838392            User 11
01925 838393            User 12
01925 838394            User 13
01925 838395            User 14

etc ...


Thank you in advance!


Phil
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060203/416de318/attachment.htm

At 11:38 AM 02/03/2006, you wrote:
>Hung up?  By whom?  Assume this: while Dial() is working (and waiting
>for the timeout) somebody has picked up a phone that shares the POTS
>line with Asterisk.  Will that second pick up of the POTS line look like
>a hangup on the POTS line to Asterisk while it is Dial()ing?
Sure seems to work that way here. I have a 4 line analog phone 
sharing the phones with * and if I grab it before a * goes to 
voicemail it never goes to voicemail.  Both my analog and SIP phones 
are ringing at the same time.

Ira