Waitexten not working as expected

Asterisk Asterisk Support
1

I wanted to get DTMF running.
Added WaitExten(5) to my extensions.conf file. Unfortunately it does not work as expected and Asterisk is not reacting on any digit.
It works in combination with Background(…) playing a sound file or if I use option d WaitExten(5,d) which send a beep tone…
From documentation I expected it to work standalone without any sound/beep/…

Is there any chance of sending the digits to Asterisk as an addition to the number to be dialed? I.e. +49246123456#1 (where +49246123456 is the number to be dialed)…
I want to avoid useraction after the call is initiated (using out of the car…)

Any ideas? Thank you very much.

2

Is Asterisk actually receiving the DTMF? Both Background and the beep tone cause Asterisk to send media. Some firewall/NAT devices won’t accept incoming traffic until this is done. For PJSIP setting rtp_keepalive on the endpoint can also allow it to work.

As for sending numbers as part of the dialed number, that’s unlikely. If the PSTN is involved then it’s really whether the PSTN would allow it and transport them. I’ve heard many years ago in some locations for some providers it did work, but not anytime soon.

If it’s a SIP client connecting to Asterisk then you certainly could.

3

Asterisk can do this, but it is less likely that your mobile operator will support this.

My guess is that either your router is not set to forward the RTP ports automatically and requires outgoing traffic, to learn the port number to open and forward, or you are behind NAT and haven’t configured the correct public media address, on Asterisk, so outbound media is necessary for the phone to use symmetric RTP to learn the address to which to send the DTMF.

4

Is Asterisk actually receiving the DTMF?
I don’t think so.
When I set logger on, there’s no activity after reaching the WaitExten command although pressing digits on the mobile. After the timeout of WaitExten(5) the log continues…

Will check with rtp_keepalive.

As for sending numbers as part of the dialed number, that’s unlikely.
This is what MSTeams for example is using if you dial in with a mobile…

5

rtp_keepalive=1
has not changed the behaviour…

6

OK, found the solution(s):

  • I had to change the port range for rtp in my firewall… Now with 10000-20000 it works
  • It is possible to send the DTMF directly with the call - I had to use the “,” after the dial number followed by the digit I want to send.
7

Germany is probably the one country where it might work. Whilst most countries send direct in dialling digits as part of a standard length number, I believe Germany uses the whole standard length number to address the PABX and send the direct in dialling digits in addition to that.

8

Don’t know about other countries but I think, that I also called into Teams sessions via mobile from outside Germany by using this technique…
With the comma the dialling is paused - so time to answer the call first.

9

That sounds like what is typically done in phones. In particular, with analogue phones, the PABX/central office decodes up to the pause, and sets up the call based on that, and the assumption is that the media path is up before the remaining digits arrive. There may be some IP phones that generate the INVITE at the comma, and then send the rest a media.

That’s basically what is being done at caller’s side, whereas you are asking about handling it at the PABX side.

I think the German DID system absorbs the whole number and passes it through in the ISDN signalling path, not as media.