GO HERE In call DTMF feature codes not working

[elihunter]

We've setup a few Polycom 650's and are having problems with dtmf. Initially dtmf wasn't working with most IVRs we dialed into. I've gotten that working but now that we're testing we've noticed in-call feature codes are not working.

I can pick up the phone and dial *97 and get voicemail so dtmf seems to work ok when not in a call.

Dialing ## and *2 do nothing while in a call, you can hear the tone play back but that's it.

I found some advice online and tried changing the following in spipm_sip.cfg

tone.dtmf.level="-9"
tone.dtmf.channel.masking="0"
tone.dtmf.rfc2833Payload="101"

I've also tried with a soft phone and ## doesn't work so I'm not sure it's an issue with the Polycoms.

Does anybody have any suggestions?

 

[elihunter]

I had the default tr in Asterisk Dial command options. I added T just in case but it hasn't changed anything.

Let me know if I'm leaving out any important information from my first post.

 

[briankelly63]

http://pbxinaflash.com/community/th...ailing-but-just-to-some-ivrs.6218/#post-37670

 

[AndyInNYC]

Sorry to revive a very old thread, but I'm having the same type of problems.

Brian, can you give a link or search terms that might work to pull up this thread post crash?

Andrew

 

[wardmundy]

Try this.

 

[AndyInNYC]

No joy.

If I call a conference service, for example, my touch tones work fine.

My problem is getting freepbx feature codes to work during a call. I have feature codes enabled for *2 and ## (along with all other default codes). If I press ## BEFORE a call, I get "Welcome to the Directory ..." which is good. If I press ## during a call or *2 I get nothing (except I hear the beeps).

This happens on my SIP phone (Mitel 5224) as well as my analog phone connected via a Digium card with hardware EC.

I can activate DND (for example) if I pick up the phone and press the digits.

Is there something 'else' I need to do to get asterisk/freepbx to recognize/act on dtmf codes during a call? Given that the results are the same on the analog and SIP phone I'm inclined to believe it is a 'box' problem rather than a phone setting - but hey, what do I know?


PIAF Installed Version = 2.0.6.2 under *HARDWARE* │
│ FreePBX Version = 2.9.0.12 │
│ Running Asterisk Version = 1.8.13.0 │
│ Asterisk Source Version = 1.8.13.0 │
│ Dahdi Source Version = 2.6.1 │
│ Libpri Source Version = 1.4.12 │
│ IP Address = 192.168.40.29 on eth0 │
│ Operating System = CentOS release 6.2 (Final) │
│ Kernel Version = 2.6.32-220.17.1.el6.i686 - 32 Bit │
│ Incredible PBX 3 Version = 3.0.9


Thanks Ward (and everyone else)


Andrew

 

[tbrummell]

Inband, or out of band signalling?

 

[AndyInNYC]

On the Mitel phone, I have set the DTMF type to Automatic and the payload type to 101 per directions I found scavenging the internet.

I'm not sure where to look on the freepbx pages/config files to see what the box thinks everything is set for. Given that both the SIP phone and the analog phone behave exactly the same, I'm inclined to think it is the box. Where do I look to answer your question?

Andrew

 

[tbrummell]

See attached, depending on FreePBX version you may need to look elsewhere in "Extensions" for it.

 

[AndyInNYC]

tbrummell,

My settings mirror yours except for my media ports.

If I press *2 or ## during the call I can (and could before) hear the beeps, but the CLI shows nothing and I don't get prompts or any 'good' behavior.

This happens from both the analog and the Mitel.

I had the sendrpid setting set to No, but have adjusted it (in the tested behavior above).

Anywhere else to check?

Thanks.

Andrew

 

[briankelly63]

I wish I had that old thread! The bottom line on it was that you want to use the rfc2833 throughout. Some phones like the Aastra 6757i have a check box in the web UI that says "Force rfc2833". If you don't do that you end up trying all the dtmf options and never getting full functionality.
Make sure rfc2833 is specifically set in the configuration for the extension itself within freepbx. If everything everywhere is set to rfc2833 and feature codes don't work then its a phone or extension config issue.

Understand that when rfc2833 is set and working properly the audio that you hear is not what is activating the feature, its a digital communication from the phone to the server. That's why when you experiment and change to inband dtmf the external IVR stuff works but then the internal functions don't. INFO was a pre-rfc2833 data communication using SIP messages. With the various options available on different endpoints they don't all handle this in the same exact way especially when they give you the option of choosing more than one signalling method.

Generally Inband dtmf is a bad idea because of the following.... would you rather send dtmf through a giant audio pipe and multiple codecs or is it better to just generate it at the very end of the pipe? The latter is the best route.

Another related tidbit for you...

http://pbxinaflash.com/community/index.php?threads/called-voicemail-delayed-response-to-key.12596/

 

[AndyInNYC]

Well, Brian, your tidbit paid off. I didn't have a T in the outbound dial. Now I do and it seems to work.
I will have to test to see if it blows anything else up, but I now receive the expected action.

Woo and Hoo!

Andrew

 

[briankelly63]

Great! Hopefully this will help clear up the whole issue. As you can see there are a lot of little things that can trip you up...