DTMF detection on bridged SIP Calls


I am not able to see DTMF events for the following call scenario.

SIP User A calls into Asterisk. The dialplan delivers the call into an AGI application

A second call is initiated to SIP User B via Asterisk AMI.
The answered call is delivered into an AGI application.

SIP User A + B are bridged together.
Whenever A or B presses DTMF I want to see the DTMF.

For the initial call into asterisk from SIP User A I can see dtfm’s both via the console and also via an Asterisk AMI login that is looking for dtmf events.

I have also verified using wireshark that the A - B call is going through asterisk from both a SIP Signalling and also an RTP point of view - asterisk is basically seeing everything.
I can also see in the wireshark rfc2833 events corresponding to the dtmf’s that I have pressed.

If the RTP’s going through Asterisk, you should see it by turning on rtp debug. Do you have directmedia enabled for the SIP peers?

Hi Malcolm,

Thanks for the timely response.

I have verified all of the RTP is going through Asterisk.
I have also confirmed directmedia is also set to no on the media gateway we are using.

I have enabled rtp debug but cannot see the rfc2833 packets in the debug that I can see on wireshark.

Any further pointers on narrowing down the issue would be appreciated.
I am currently looking at the asterisk code to see if I can pinpoint the issue.


Is wireshark generating a capture from the switch port or from the interface on the machine?

the tcpdump pcap is being generated from the Asterisk box.

I can see all the RTP Event’s arriving into Asterisk and being transitted out.

It’s probably packet to packet bridging. You need to enable a feature that is incompatible with directmedia, not just turn directmedia off.