[asterisk-users] Debugging DTMF

[Adrian Marsh]

Hi All,

 

I'm trying to debug DTMF issues I have with certain endpoint
conferencing systems (external, 3rd party).

 

On our A*k server I log DTMF, and I see that coming through in the log.

What I'd like to see is what is sent onto our VoIP carrier over SIP.

 

I can do a tcpdump of the packets, but what am I then looking for?
Would it be in the RTP audio stream or within the SIP protocol??  I'm
using Wireshark to decode...

 

Thanks,

 

Adrian

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