I want to replace the analog trunk lines of an old PBX with an Asterisk PBX.
The old PBX is a Doro/Nitsuko DX-E.
Since this is in Sweden I’m not sure if it is E&M wink start or
something similar. The PBX reseller is currently searching their
documentation for this.
The Telco doesn’t want to support analog DID anymore, so I thought
Asterisk might help keeping the old PBX alive for a little longer.
I have found the Asterisk Dial command option “D([called][:calling])” :
“Sends DTMF digits after the call has been answered, but before the
call is bridged.” But no corresponding option for pulse sending. But
this may be the wrong place to look, I don’t know. Could it be as
simple as just doing a standard Dial(…) command?
I think a TDM400 or TDM2400 with FXS modules can do the trick, but
since I haven’t found any configuration documentation I’m stuck.
And I also found, for the “pulse=yes” to work: "…you can apply a patch file to enable you to specify pulse dialing with the pulse keyword. "
I’m still not sure if it really is E&M wink start. When I know that I will get a TDM card and try it out. For the momemt I have only tested with a DVG-1402S (two channnel SIP to FXS). It understands pulses but does not seem to be able to produce any.
One big difference in the signalling is that the american standard uses Loop Reverse-Battery Signaling, the swedish does not. (This is which part that provides the telephone line voltage.) There are more differences, I’m still reading the documents.
I guess what Asterisk does is according to the american standard.
I also begin to suspect that this forum is wrong place to tell this story, no one else here seems to have tried this before, I feel like I’m talking to myself.
You are right, the easiest/fastest way forward is to kick out the old PBX.
And that is going to happen sooner or later.
In fact, the solution to “the PBX problem” was “buy a new PBX”.
But when I brought up the idea of using Asterisk I got positive response,
especially when I said it will be quite a bit less expensive.
I was close to buying a TDM400 after having read about pulse dialing on
the voip-info wiki, but after finding the signalling standard I hesitated.
For the moment I am looking at the details, to see if it really is possible
to do analog DID.
My situation doesn’t EXACTLY parallel yours, but this I can relate.
The TDM400 FXO WILL pulse dial WITHOUT any patching. E&M wink doesn’t seem to relate to any but the T1/E1 cards
What several us are doing is using an FXO card to pulse dial into either an incoming selector or station line of electromechanical switches. ( so that makes your PBX a youngster )
We have a private network of interconnected Asterisk boxes, a specialized ENUM DNS reference point that allows dialing between these antique switches. US, UK and NL collectors are all participating.
A few "gotchas"
Pulse dialing on the FXO most probably needs to be loop rather than “kewl”. Any reversal of the loop from your PBX probably will drop the call, though there are options that you can try to change that.
the “w” in the dial string DOES NOT work in pulse. Under the sheets Asdterisk plays a sound file of silence when it sees the w, so in pulse it doesn’t even pause!
IF you are to use an FXS module, also be aware that the Grount Start, or Earth Calling function seems to have stopped working around the introduction of 1.2 stable.
It seems to still work in the trunk versions though. This isn’t directly related to what you are trying to do.
Finally, really consider the Sangoma A200 card. Somewhat less cost, here in the US, works with MANY more motherboards, and their support, at least via phone is good.
I found a defect in their FXS driver in pulse dial that should be in the current, though still beta, release.
Since I’m testing Asterisk on the computer I found at the bottom of the
trash pile in the computer room, the fact that the Sangoma card works with most motherboards was interesting. (I intentionally use a slow CPU to see when Asterisk gives up.)
I read the Sangoma documentation and successfully compiled and installed their drivers.
Now I have orded a Sangoma A20100 (two FXS) to see if it really works.
The Sangoma A20100 (two FXS) card is working with the old test omputer (it has a 333 Mhz CPU).
I first tested with an analog telephone, and then connected to a spare line of the old PBX. The PBX answers a call from Asterisk and the other way around also works (ie, a dial tone and dialing out via Asterisk).
But I still don’t know if the DID part works. I can’t disconnect any of the (DID sensitive) inward directed lines for testing without disrupting normal service. Now I’m waiting for the PBX service technician (currently on vacation) to help me reprogramming some more lines for DID.
I wrote a small extension to the zaptel routines for handling
decadic signalling. And it has been up and running all of this year.
Ie, an Asterisk calling into the old PBX. The old analog lines
are replaced by SIP, but the old PBX is still fully functional,
just dial an extra zero to get through the Asterisk PBX.
I have submitted the source code additions to the zaptel Digium
group (this summer) but I don’t know if it will get accepted.
however i have no answers for you except it maybe time to replace that old PBX afterall and go totally asterisk