Scenario: Customer has an old PBX. They dial 9 to get an outside line that was connected to a T1 provided by the PSTN. After dialing 9, the user would hear a new dialtone and all digits dialed would go straight out the T1 to the PSTN.
Now, replace the PSTN T1 with a T1 to an asterisk server. The user still dials 9 and the PBX still connects the caller directly to the T1 that now goes to Asterisk. But now the user does not hear a dialtone and does not know to dial numbers.
Asterisk will be sending the call out over a SIP trunk once it gets the digits to dial.
Question: Is there a way to make Asterisk’s T1 act just like the PSTN’s T1?
Create an outbound route that points to your T-1 and in the dial patterns put:
That does it for PSTN lines.
Thanks for the suggestion. The PBX does not send Asterisk the 9, it expects to just pick up the PSTN T1, then the caller should hear a dial tone and dial digits. Since the call is no longer going directly to the PSTN over the T1, but rather goes to Asterisk, then to a SIP trunk, how will the caller hear dial tone?
PBX -> T1 -> asterisk -> SIP trunk -> soft switch -> PSTN
Thanks for the help.
if I’m understanding you correctly:
[OLD-PBX] --> [Asterisk] --> SIP_or_othre_PSTN_Connection
Sounds like you need to have the call immediately transfered to a DISA connection on Asterisk. In the case of an FXS this would be setting immediate=yes with a context defined that dropped the channel into DISA. For a T1 I suspect there is a way to do the same, you’ll have to poke around a bit to figure out how to fire off the call.
It may just be context=my-disa-dialtone in the zapata.conf file and then in extensions_additional.conf
exten => _.,1,Disa(no-password|from-internal)
exten => _.,n,hangup
assuming you want the dialed calls to drop into from-internal (giving them access to all internal extensions and features as well as all outbound routes).
That’s a great use for DISA! I was considering a custom context, but had not figured out how to match a pattern for outbound dialing. I’ll give it a try and let you know how it turned out.
I’m having same problem. I need to generate dial tone on T1 non-PRI d4, ami, e&m wink
I was trying to use DISA but I don’t find the way to get into it without dialing any number first.
I think the problem of the second dial tone is overrated. When I put in our 1st asterisk system on a PRI, I nearly panicked when I realized that there was no second dial tone after dialing 9. I then realized that you didn’t even have to dial a “9”.
So, when I commissioned the system, as part of the training I simply told the users that “You no longer have to dial 9. But if you do, It’s OK.”. Took our users all of 71 1/2 seconds to embrace the new procedures.
In other words, use the old marketing trick. Turn something that might be perceived as a negative (you don’t get a 2nd dialtone after dialing 9) to a positive (but, you don’t have to dial 9 anymore).
Of all the features Asterisk/FreePBX provides, I think I got the biggest positive feedback from the users not having to dial “9”. Go Figure!
The problem is that I’m connecting a dialer via T1 that need dialtone to be able to dial on T1 Wink and I don’t find how to make this work.