Ouch. This is getting ominous. I see what you mean. The DID lines would 
be a form of FXS line.

I thought I saw mention of a conversion box somewhere.  Maybe I'm 
thinking of the one for the proprietary phones though.  There must be a 
solution to this, other than PRI I mean.  The weird thing is that there is ring 
voltage when the call comes in on the DID trunk. Since we are supplying 
the voltage to the line, I guess I have to assume that the ring is coming 
from our equipment.  This definitely got me started on the wrong tangent for 
the project so I'm not even sure how the DID is supposed to work

Thanks for the input though.  I'll be wracking the old brain today. E&M Wink 
start only applies to T1s right?

Regards,
Peter M.

> If you're talking Analogue DID, then I don't know if anyone'e ever done that
> with Asterisk/Zaptel.
> 
> What you'd need is not to use FXO, but rather FXS, since the PBX is
> responsible for providing battery (48V) on an analog DID. Also, I'm not sure
> if there's a standard with respect to immediate/groundstart/delay dial, but
> that'll have to match.
> 
> Finally (and possibly most problematic) is that the FXS will deliver dial
> tone, which the C.O. will NOT be expecting. This may or may not cause a
> problem, and I am not sure what a neat way to turn off the dial tone would
> be (some nifty hacking in indications.conf would be a start, I suppose, but
> you're going where no one has gone before, so be prepared to blaze a trail)!
> 
> Jim.
> 
> --
> Jim Van Meggelen
> [EMAIL PROTECTED]
> http://www.oreillynet.com/pub/au/2177
> 
> "A child is the ultimate startup, and I have three. 
> This makes me rich."
>                     Guy Kawasaki
> --
> 
> 
> > -----Original Message-----
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] 
> > Sent: February 8, 2006 9:06 AM
> > To: [email protected]
> > Subject: [on-asterisk] DID lines
> > 
> > Just wondering if anyone has run into the situation with 
> > trying to connect DID trunks to an Asterisk server.  We've 
> > run into an impass where Aliant can set up the loop as pass 
> > the calls but they can't transfer the digits of the end 
> > number on the loop because this is "technically not 
> > possible".  Is this true?  Do I need to look at a DID to 
> > loop-start converter or something?  Or would there be a * 
> > compatible DID card available?
> > 
> > Thanks,
> > Peter M.

********************************************************
Peter MacFarlane, ACP
Network Administration &  Programming     
Target Call Center/ Message Centre P.E.I.  
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