Hi. Many thanks for your email. I'm using fusion PBX. Is that still based on asterisk's method of configuration? Very best wishes, Mo. On 07/06/2017 02:05, Jason White via Blind-sysadmins wrote:

Mobeen Iqbal wrote:

...

Does anyone have any idea what I need to do to get in call codes working?

First question: is your Asterisk installation recognizing DTMF tones? If you try voicemail or another feature that requires them, are they properly detected?

There are different settings for sending/receiving DTMF if you're using a SIP phone, so make sure that the client side is properly configured. As noted on the following Web page, DTMF can be distorted by audio compression, which is why you should use RFC2833 or one of the other standards mentioned: https://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode

If DTMF is working, then the problem will be somewhere in the Asterisk configuration. It's a long time since I've dealt with Asterisk configuration, so I'm probably not going to be of much help.

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Bill Dengler wrote:

No, it uses FreeSWITCH.

That's good - I can help a little with FreeSWITCH, if necessary. However, I suggest following Bill's advice and trying the IRC channel first. The default FreeSWITCH configuration (the one supplied with FreeSWITCH itself) does include the commands that you want, configured using the bind_meta facility - see dialplan/default.xml and dialplan/features.xml for the actual feature configurations.