I’ve been trying to figure this out, but not much luck. Trunk is set to auto, and the extensions are (now) set to inband (have also tried RFC2833 on both trunk and extensions). In particular I’m working with a Polycom IP7000 Phone (if that makes any difference). Calls go out over a Sangoma A400 card.
I’ve also tried disabling hardware echo cancelation on the card (no improvement).
The specific behavior is that when trying to enter the conference codes (for external services) on the phone we get doubled digits, so they can’t get in. Using super fast touches seems to increase likelihood of success somewhat.
Freepbx 13.0.192.19 on a Sangoma 50 box. (Upgrades are stuck and normal unlock isn’t fixing that either).
Who is your conference provider? If they offer SIP access, try that. In addition to solving your DTMF issues, you’ll likely get much better voice quality (assuming that you’ve solved your internet stability issue).
If you have any input for choice of conference provider, consider https://www.freeconferencecallhd.com/ . You get a wideband (G.722) connection via SIP or their mobile app. Though not advertised, you can connect to a SIP URI like [email protected] , where 7127704160 is the access number and 123456 is the Host or Participant code. This brings you directly into the meeting without having to enter the code manually. If you don’t need FreePBX to log or record the call (you will still have the recording from the conference provider), you can set up the SIP URI as a speed dial on the 7000 – press just 3 keys to access your meeting.
If you are stuck with a conference provider without SIP access, you may still get better results using a SIP trunk for the call; that eliminates the gratuitous D/A and A/D conversions and echo path associated with the analog line.
You might also look into the DTMF keypress duration, which can be programmed in some phones. Cisco phones (which you aren’t using) has the capability to tune the DTMF press settings to specific durations.
No SIP option for this: the ISP in this building is terrible with regular momentary drops. We had to go to analog to get better call quality and reliability.
“In call” DTMF signalling over either DAHDI TDM (PRI’s and legacy T1’s) or PSTN (ANALOG) can ONLY ever use INBAND , any bridged calls over SIP, IAX2 or whatever, should never try and negotiate anything else. However, some very old landline providers do need a longer DTMF.