How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no messages related to DTMF... or if I just do a global SIP debug for that matter.... I am using RFC DTMF but it's not being passed to the PSTN and I need to debug this further. I've tried to increase the verbosity and the debug ('set debug n') and that didn't help either. I assume this is because even RFC2833 sends the DTMF as RTP which wouldn't show up anyways.... but how to troubleshoot DTMF issues?
On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:> How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no > messages related to DTMF... or if I just do a global SIP debug for > that matter.... I am using RFC DTMF but it's not being passed to the > PSTN and I need to debug this further. I've tried to increase the > verbosity and the debug ('set debug n') and that didn't help either. I > assume this is because even RFC2833 sends the DTMF as RTP which > wouldn't show up anyways.... but how to troubleshoot DTMF issues?I'd first turn on "rtp debug" and see if that helps. If that's not enough information, I'd go into logger.conf and add "dtmf" to the logger and messages lines (and any others you care about), and then do a "logger reload" from the Asterisk CLI. -- Jared Smith Community Relations Manager Digium, Inc.
I think your best bet is to do a packet capture and look for RTP packets with an RTP Event payload ("rtpevent" display filter). On Mon, 28 Jan 2008, Andrew Joakimsen wrote:> How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no > messages related to DTMF... or if I just do a global SIP debug for > that matter.... I am using RFC DTMF but it's not being passed to the > PSTN and I need to debug this further. I've tried to increase the > verbosity and the debug ('set debug n') and that didn't help either. I > assume this is because even RFC2833 sends the DTMF as RTP which > wouldn't show up anyways.... but how to troubleshoot DTMF issues? > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671