DTMF detection on bridged SIP Calls

Asterisk Asterisk Support
1

Hi,

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.

2

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?

3

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.

Thanks,
Neil.

4

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

5

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.

6

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