DTFM not working working when call is established


I am playing around with Freepbx for evaluation purpose i used raspbx.

A simple setup with sipura ip phones and chan dongle.

The only thing that doesnt work is using the “dial number for” menues.

I only tried voicemail. that didnt work. enabling debug logging for dtfm
brought nothing. like i didnt press anything. does not matter if internal (voip->asterisk)
or external (gsm phone → chandongle → asterisk)

any hints. must be something simple for i didnt change much from the default config.

any help would be nice

Your codec is ?
Your dtmfmode is?

where do i find that information?

the freepbx help function does give no result for either.

Do mobile networks actually support DTMF towards the subscriber? I can’t think why they would, as they were intended for voice. They have to handle it towards the network, but that is done out of band, with the inband DTMF being generated within the fixed network.

One FreePBX ‘help function’ is the Wiki , which is linked to at the top of the page you are reading.



ok i found the codecs under:
Settings → Asterisk SIP Settings → General SIP Settings
there is in that order and activated:
g729, ulaw, alaw, gsm, g726, g722 ,g719, slin

didnt find a DTMF settings.

edit: on the used sip phone i selected prefered codec: g729a
it also supports: g711u, g711a, g726-16, g726-24, g726-32, g726-40,g723

shure dialing through menus works via mobile phone. im doing it all the time.
just havent managed to get it going with freepbx yet :slight_smile:

There is also the Search help function included with freepbx. it works quite well! (top left, the magnification/search symbol) (just for finding topics, not items)

g729 cannot support inband dtmf, choose a g711 varient for vanilla.

Yep, that box searches the self same wiki

mhm, ok, choose it on the phone. but with freepbx there is no such codec in the list

the same codec g711a g711u are also known as ulaw and alaw (its in the wiki under g711)

ic, see, now that is unintuitive

so is typing a μ

ulaw is used. no working voicebox menu.

It might be allowed , but during the call, is it being used , during an active call

sip show channels 

shows a format column, if is in fact ulaw and you have dtmf logging enabled on the ‘console’ log, then . . . .?

while i listen to the mailbox menu: keypresses echo locally on the phone but are not registered by asterisk.

sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
0 active SIP dialogs
– <PJSIP/101-0000004f> Playing ‘vm-messages.ulaw’ (language ‘en’)
– <PJSIP/101-0000004f> Playing ‘vm-opts.ulaw’ (language ‘en’)
raspbx*CLI> sip show channels

now experiment with the dtmfmode, I have no experience with chan_dongle, but if using inband, you should at least hear a click, both RFC2833 and inband might struggle with it’s channelization, so maybe ‘info’ , where the SIP protocol carries the info.

GSM 03.14 version 5.0.0 only provides protocol details for DTMF going away from the mobile station, and section 4 says: “The support of this facility in the land to mobile direction is for further study.”