DTMF signaling not working with SPA942 phones with Sangoma AFT-200

Freepbx 12.0.57 with latest updates.
SPA942 SIP phones with latest firmware.

Users are getting invalid keypad tones with attempting to walk through another companies Automated response systems. The tones sound strange, not giving the correct number tone when asked to press a keypad number to walk through the menu options on other companies phone systems.

Checked log file…nothing out of the ordinary.

Started running Freepbx on Friday 4/17/2015 and rest of the system is a-ok.

Any ideas?
Thanks for the help.

The issue is with DTMF signaling setting.
Set to default of rfc2833 in Extensions settings.
Changing Extension option to Auto for 1 extension but it did not fix the problem so set it back to rfc2833.

More testing done.
Pressing the number keys in response to an external IVR request cause multiple tones to be sent for the same number key.
Fast or slower key pressing does not make any difference.
so for instance, press the 7 key once and you hear (tone, tone …or tone, tone , tone) in rapid succession. The external IVR interprets this a duplicate key pushes.

I was wondering if the problem is related to echo cancellation?
I have a Sangoma ATF-200 4 port analog.
What should the echocanceller option be set to?
mg2 or oslec?

Any help would be appreciated.

I just ran into this today on these phones - latest firmware 6.1.5a loaded. The only fix is to select the uncompressed codec g711u and select inband DTMF signaling in the extension menu on the phone. While the phone does “support” rfc2833 it’s obviously buggy.

I have several SPA942 running on latest firmware with G711u codec and RFC2833 in combination with Sangoma FXO cards and none of them present any issues with DTMF signalling.

I have no problems with dtmf signaling to the voicemail, etc. within FreePBX when the phone is set to use AVT or INFO. The problem is only sending DTMF to remote IVRs at banks, etc. It doesn’t happen with any other model of phone I have tried (Yealink, Polycom, etc.) on my system.
This model phone has been out of production for close to a decade and their value on the used market is effectively zero (I bought my sample for $5 for example) so this is more historical interest than anything else. (In my case I only have my 1 sample phone all others are different models) I use a FXO gateway not a card in the machine running FreePBX so that may have something to do with it. Note that the Advanced properties in the Extension have to be configured to use inband signaling, not just the phone. Both need to be changed.
DTMF was designed to be carried inband in the phone system, the dual-tone frequencies were selected so that a human couldn’t imitate them. What most don’t understand is that the out-of-band signaling is only in force within the phone system - in between the phone and the gateway to the PSTN. Whether that gateway is at the site (like on both of our cases) or in “the cloud” (such as if your trunks are SIP trunks) when the call reaches a FXO port and is put out onto the PSTN the FXO port converts the out of band signaling into DTMF that is put inband on the PSTN.
The only reason it ever became an issue with VoIP is because of the use of compressed codecs which are only of value on a network that has low jitter such as within a LAN. So, while the “purists” may turn up their nose at in-band signaling of DTMF the reality is that inband signaling is the “normal” way the “real” phone network handles it.
This factoid brought to you by the Holy Jihad War Crusade Against The Wrong Way To Do Networking.

I can confirm that at least in my case, all my SPA942 phones are working correctly in regards to RFC2833 DTMF tones sent to the outside, either on g711 or g729, with either FXO cards or FXO gateways. I understand it might not work for others, but fortunately they do work for me.

As I said DTMF is encoded only within the phone system, once it goes out a FXO gateway or card, it’s converted back to tones. You stated you used Sangoma FXO cards, but you did not say what model gateway you have used. In my case I’m using a Cisco 1700 series router with voice ports.

Assuming Asterisk is not re-coding DTMF then what you are saying is merely that SPA942 dtmf-to-Sangoma FXO card is working, what I am saying SPA942 dtmf-to-Cisco 1700 is not. Without saying what gateway or gateways you have used (as opposed to Sangoma FXO cards) you really aren’t adding anything new here

This isn’t an issue of it “not working for others and works for me” It is an issue of it works with some combination of gear and not others. You got lucky to get compatible gear but the fact it works for you has nothing to do at all with any configuration you have done. It’s most likely due to at some point in the past Sangoma happened to test a 9xx series with their card and realized they needed to put some kind of workaround in for Sipura brokenness in that firmware.

(Sipura is the original author of the code in that phone, BTW)

I was just trying to contribute. The systems in question have a sangoma b600 in one of them and a grandstream gxw4104 on the other, so one has a card, the other has a gateway and both of them have spa942 phones working with dtmf on rfc2833.