[VoIP] Mitel Smart-1 Pulse to DTMF converter

Donald Froula dfroula at sbcglobal.net
Wed Oct 25 20:43:23 CDT 2006


My thought is that since I will be using the Mitel in
front of the PAP2, the store and forward operation
would be acceptable, since the ATA is already doing
much the same before initiating a SIP session to
Asterisk, with the dial plan I have programmed into
it. I don't believe the Mitel passes through the dial
pulses as the Sandman unit does, so false supervision
detection should not be an issue.

Don

--- Steph Kerman <stfkerman at jps.net> wrote:

> One consideration in the use of a Smart-1 for pulse
> conversion in 
> comparison to other options is that the Smart-1 is a
> store & forward 
> dialer.  During the time it is acquiring digits and
> until its dialing 
> plan programming has been satisfied so that it can
> outpulse, you are 
> split by the Smart-1 from the downstream switching
> system. 
> 
> Any response that might come from the switching
> system before a full 
> number has been dialed, such as a response to
> dialing a vacant code, 
> will not be heard until the Smart-1 has received all
> the digits it 
> expects and has been satisfied.  For the same
> reason, you would not be 
> able to use it for end-end signaling even in a
> context that did not 
> require the # and * keys.
> 
> These issues may not be a concern in a given
> application.  That is for 
> you to decide.  In principle, a dedicated pulse
> converter would convert 
> the digits on the fly and not present this issue. 
> Sandman makes such a 
> unit but I believe it has a design problem that have
> been discussed here 
> or on the TCI list in the recent past.  My
> recollection is that it 
> allows the dial pulses to pass to the DTMF or
> "switch" side of the 
> connection. 
> 
> Some ATAs do not accept dial pulses but do not
> ignore them either.  They 
> see each dial pulse as a disconnect signal.  The
> Sandman unit will not 
> work with such ATAs because each digit it
> interpreted as a series of 
> on-hook/off-hook transitions.  However if the ATA
> disconnect timing is 
> long enough to ignore dial pulses, the Sandman unit
> may be a better 
> choice than a Smart-1.  It's also probably feasible
> to add some 
> relatively simple circuitry between the Sandman unit
> and the ATA to 
> prevent the dial pulses from reaching the ATA.  That
> circuitry would 
> need to be hand-built or accomplished by modifying
> an existing stock 
> device.  I can discuss this in further detail if
> there is interest.
> 
> Steph
> 
> John Novack wrote:
> > Doug Alderdice  is the Smart-1 in this arena.
> > Did he ever get his power back?
> > Used Smart-1's need to be thoroughly checked,
> since older ones had NiCad 
> > batteries that leaked and ruined the unit.
> > The Sangoma A200 card may be a better bet in the
> long run
> > I have just finished a session with them on
> getting them to make pulse 
> > dial work properly on their A200 card.
> > Not the lowest cost solution, but a good one
> > For six FXS ports you would need one expansion
> board.
> > Break even for T1 is probably at 8 ports,
> depending on what you pay for 
> > a used channel bank off eBay
> >
> > John Novack
> >
> >
> > Donald Froula wrote:
> >   
> >> I have a fairly large collection of rotary phones
> that
> >> I would like to use on my Asterisk system. I am
> using
> >> three Linksys PAP2-NA adapters to connect 6
> phones to
> >> the switch. I have been using the voice
> recognition
> >> module mentioned here and the PAP2 in "bat phone"
> mode
> >> with my rotary phones.
> >>
> >> I just purchased a Mitel Smart-1 box to act as a
> >> rotary to DTMF converter. Would anyone have any
> >> programming tips to set up the Mitel to work as a
> >> simple converter?
> >>
> >> Don
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 




More information about the VoIP mailing list