Operation of an IVR on the other side does not work

FreePBX Configuration
1

Hello
I have installed FreePBX 16.0.40.7 on a Raspberry Pi 4. I use IVR for incoming calls myself. When I make outgoing calls myself and encounter an IVR on the other side, it doesn’t work.
So if I type in menu numbers on my phone, it has no effect on the other side’s IVR.
How do I have to configure it so that I can successfully enter IVR numbers during a phone call? I suspect that this could have something to do with the outbound rules, but I’m not sure. Do you have any ideas or suggestions on this?

Best regards

Niclas

3

Hello Bert,
and thank you for this detailed how-to. I’m currently testing it. But I’m sure I can get it to work.
Best regards
Niclas

4

Well, it doesn’t work quite as smoothly as that. Because in the Asterisk CLI I just discovered that “dtmf” doesn’t exist:

> dtmf set debug on
No such command 'dtmf set debug on' (type 'core show help dtmf set' for other possible commands)

and

> core show help dtmf 
No such command 'dtmf'.

I’m also looking for DTMF among the installable modules in the GUI, but I can’t find it.

5

That CLI command doesn’t exist in Asterisk, and never has (yet another case of AI making things up). In normal Asterisk DTMF logging is configured in logger.conf for the respective channel (file, console).

7

(So, hopefully you can see the problems involved in starting a venture based on AI acquired word salads (inferred from your other post), my suggestion, spend a couple of hours doing the old fashioned Reading of the FMs first :wink: )

9

No, as far as I know a Bit Error Rate Test in the Telephony world is not AI but an electrical analysis of the quality of a digital signal on a digital trunk as received at the far end of the loop.

Or did you assume Bert was something else?

10

For anyone experiencing the same problem:

DTMF Behavior on Outgoing Calls

For outgoing calls, DTMF (Dual Tone Multi-Frequency) signals can be used to select menu items in external IVR systems. In certain configurations, keystroke commands (e.g., “1,” “2,” “#”) are recognized locally but do not always reach the other party correctly.

This behavior is generally not related to the outbound routes, but rather to the DTMF transmission configuration in FreePBX or Asterisk.

DTMF Transmission Types

Inband
DTMF tones are transmitted directly in the audio stream. Only works with uncompressed codecs such as G.711.

RFC2833 / RFC4733
DTMF tones are transmitted as separate RTP packets.

SIP INFO
DTMF tones are transmitted as SIP messages. This method is supported by some providers, but is less common.

If FreePBX/Asterisk, the device, or the provider use different methods, this may result in DTMF signals not being correctly forwarded to the other party.

Connectivity → Trunks → Trunk → Tab „pjsip Settings“ → Tab „Advanced“

“rfc4733” is usually the right choice!

11

I’d say that in most cases G.711 is ALSO the right choice! The only time I’ve ever gotten good sound quality out of the compressing codecs is when the circuit they were run over was highly controlled and not lossy, or had enormous amounts of bandwidth (like a gigabit switch fabric)

64kbps today is miniscule compared to the typical Internet circuit. It’s hard to believe but I still remember installing 56k dedicated circuits and 56k DSU’s as that was considered an acceptable circuit speed “back in the olden days” LOL

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