Does anyone know how you can configure freePBX so you can dial 0 and hear the external dial tone?
just checking because im trying to recreate a true 1980s mechanical PBX
thanks B.
Does anyone know how you can configure freePBX so you can dial 0 and hear the external dial tone?
just checking because im trying to recreate a true 1980s mechanical PBX
thanks B.
What capabilities of FreePBX do you want to retain after the simulated external line?
Howdy @ringedseal13 - Welcome to the forum and thanks for using FreePBX!
What are you using as the dial out? Do you have a SIP service or are you trying to grab a POTs line from say an FXO port?
just to dial between extentions. ![]()
Hey! @mwhite
Im using the Grandstream Ht812v1 It has no FXO and I just wanted to see if there is anything I could do to simulate this
Thanks.
If you’re trying to replicate an old hardware PBX they never required you to pick up an “external line” to call internally. If you wanted to reach 200 you just picked up the phone and dial 200 to reach 200.
First, you will need to accept the fact that the ATA, FXS gateway, IP Phones are all going to provide dial tone locally to the device. They pick up the handset connected to the HT812, they will have dial tone.
Second, you will have to reprogram the HT812 to make sure the dial pattern in the device itself accepts a single digit and immediately sends it. No device has a dial pattern that matches a single digit and sends it immediately. Right now they would press 0 and the HT812 would wait until its timeout before sending if it accepts a single digit.
Third, you will need dialplan in FreePBX that will accept 9 as a single dialled digit. That dialplan will need to mimic an IVR since Asterisk will need to answer the call to do anything like take more digits. This includes playing back dial tone sounds. So you basically need an IVR option that matches the outbound dial patterns for your calling area and sends the call to the outbound routes (or where ever else)
Fourth, depending on what you’re really trying to do you’re going to have to lock down FreePBX side with some sort of custom dialplan. You’ll want to make sure the devices like the HT812 will only accept and send the digit patterns you want.
So use picks up phone, has dialtone, dials 0 for “outside line” which sends 0 to the PBX and the calls is answered waiting for DTMF (only way to accept the digits that follow), use enters number, DTMF matched on PBX and calls sent to outbound routes or where ever you want.
– You will need to make custom dialplan
– a custom context that all the extensions can use
– update all extensions to use that custom context
– update dial patterns in devices to accept and immediately send 0
Please give a specific example of what the user dials and hears when calling the other port.
Possibly, the HT can do what you want, without a PBX.
Without doing anything, you can dial ***701 to call the phone on FXS1, or dial ***702 to call the phone on FXS2.
In
take a look at “prompt tone”. Unfortunately, the manual doesn’t say what happens after you access the prompt tone and hear it. If it continues processing the dial plan, you should be able to set it up so the desired input sequence rings the other phone.
If doing it with a PBX, see
(old document but I believe it still applies).
@ringedseal13 Without a real analog line, you can’t present a genuine dial tone, but you can get pretty close with some custom dialplan logic. For example, the user dials 0, hears a simulated dial tone, enters some digits, and FreePBX routes the call out a SIP trunk.