This is very old technology, and it may be better to use a more modern
approach but.....
I would imagine that if the FXO port is capable of E&M - wink start you
might be able get this up and running.
For * you need to set this up as a DISA (Direct Inward System Access) line
The trick will be to delay the dialled digits (last 3 or 4 of the
number) after the answering the call
(you will need a few millisecs for the trunk to settle) and then passing
these digits as an extension number.
I "DID " one of these about 20 year ago(Mitel SX100) and I remember
what a pain it was to get the timing right .
If you want to save a few hours use a butt-set on the trunk to monitor
exactly what is occuring
First: get the trunk to answer the call
Second: signal the trunk that you want the rest of the digits dialled
Capture the digits and use as an extension number (supply ring back tone
etc.)
Good Luck
Henry
Dave Donovan wrote:
Hi Peter,
If I understand correctly, this is the sequence of events:
1) Aliant sends you ringing
2) Your system should answer
3) Aliant sends you DTMF tones indicating the DID (like ISDN DNIS)
4) You collect the digits and route the call playing ring tone back to
the caller
If that's correct then it sounds like you just need an autoattendant
script without the recordings. Answer the call and setup an extension
for each set of DID digits that you could receive.
Your message says you're trying to get away from this wink-start ->
tone-burst situation you've got now. Is that because asterisk can't
handle it or for another reason? If asterisk can handle the
wink-start, then it seems like you might have a pretty simple solution
by just letting asterisk capture the tones.
Correct me if I've misunderstood something.
Dave
On 2/8/06, [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>
<[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
Thanks Jim. You can see how weird the situation is then. What
we're trying
to do is fairly basic I think. Instead of our ancient PBX
answering the wink
start line and receiving the DID digits as a tone burst, we want
the same
thing coming into a loop start FXO card instead, only ring voltage
from the
CO indicating an incoming call first. They got that working after
two trips to
the design dept. Apparently this falls down where they are unable to
program the virtual phone number to be transferred in place of the
CID info.
* is answering the call though. So close and yet so far away.
I found this article when seem to indicate that ISDN is a
substitute so I'm
waiting on an assessment of that option. Now I just need to find
a multiport
ISDN card.
http://kbase.gfi.com/showarticle.asp?id=KBID001349
We would go to a PRI but the cost is prohibitive for the level of
messaging
services we are doing right now. Maybe when we look at the call
centre
side, we will have PRI. Reality is we will probably need to
support some
hardwired lines for a few clients, but the DID line problem seems
to be
another kind of beast.
Next time I get a call from Aliant tech support, I think I'll ask
if its "Al"
calling. That anonymous caller business is stupid.
Peter M.
> What is your application? If we know what you're trying to
integrate, we
> might be better able to brainstorm it with you. Analogue is
generally a bad
> idea when you need complex signalling, and the costs add up fast.
>
> --
> Jim Van Meggelen
> [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
> http://www.oreillynet.com/pub/au/2177
>
> "
--
Henry L. Coleman
www.voip-pbx.ca <http://www.voip-pbx.ca>
1 866 415 5355
"The Future Is Not What It Used To Be"
