Dtmf signaling not working pjsip extensions

Guys,

I’m having no luck what so ever getting external IVR’s working with my phone set up.

I have been mucking around with the various dtmfmode’s for ages and am getting no where.

I currently have everything, including the trunks set to rfc4733 and have run asterisk -rvvv and sip set debug on but can not see anything in the log to indicate that the dtmf is even being set out.

I can see the call establish a simple bridge, but that’s it.

---
    -- SIP/NetFone-0000000a answered PJSIP/2010-0000000a
    -- Channel SIP/NetFone-0000000a joined 'simple_bridge' basic-bridge <0f77de69-aab9-4708-bce1-9bc43f5df66d>
    -- Channel PJSIP/2010-0000000a joined 'simple_bridge' basic-bridge <0f77de69-aab9-4708-bce1-9bc43f5df66d>
Reliably Transmitting (NAT) to 203.2.134.1:5060:
OPTIONS sip:sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP 150.101.1xx.xxx:5061;branch=z9hG4bK3742fdc5;rport
Max-Forwards: 70
From: "Unknown" <sip:0390362yyy@150.101.1xx.xxx:5061>;tag=as4dc98e64
To: <sip:sip.internode.on.net>
Contact: <sip:0390362yyy@150.101.1xx.xxx:5061>
Call-ID: 3b2be87368941cca649954cc33948a56@150.101.1xx.xxx:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-13.0.190.2(13.9.1)
Date: Wed, 09 Nov 2016 05:47:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 125.213.160.81:5060:
OPTIONS sip:sip00.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 150.101.1xx.xxx:5061;branch=z9hG4bK12761346;rport
Max-Forwards: 70
From: "Unknown" <sip:09103xxx@150.101.1xx.xxx:5061>;tag=as0077b0bd
To: <sip:sip00.mynetfone.com.au>
Contact: <sip:09103yyy@150.101.1xx.xxx:5061>
Call-ID: 594ad05426afffa76d82c5f069e21732@150.101.1xxx.xxx:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-13.0.190.2(13.9.1)

and nothing else, no matter what keys I press on the phone until I hang up.

Any ideas where to start looking for this?

The system is an AsteriskNow install, using Asterisk 13.9.1 if that helps.

Peter.

Please provide the SIP logging for the INVITE transaction.

Also please provide the RTP debug output, as the preferred method for sending DTMF is RFC 2833 (RFC 4733) which uses specially coded packets in the media stream not ones in the signalling stream.