Martin Smith
2008-Jun-09 20:36 UTC
[asterisk-users] RFC2833 DTMF -- with an RTP debug log -- need some analysis/interpretation
Hello all, I've got an Asterisk system I'm working on here, and we often dial remote IVR systems, where our end must enter an extension to get to a remote user. We're using Polycom hardphones here, speaking SIP, and Asterisk sends these out over a PRI line with Zaptel hardware. I've used rtp debug on the phone, and I've got output, but I can't tell if it's correct or not -- I was dialing extension 221, but the remote system lost one or more of the digits. I'd appreciate another few pairs of eyes checking out the rtp debug... [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '2' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 60 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end accepted with begin '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' has duration 60 but want minimum 80, emulating on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end emulation of '2' queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin '2' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin ignored '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 60 ms [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' has duration 60 but want minimum 80, emulating on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end emulation of '2' queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 222 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' put into dtmf queue on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin emulation of '2' with duration 100 queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '1' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '1' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' received on SIP/199-b31ddc00, duration 80 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' put into dtmf queue on SIP/199-b31ddc00 Thanks! Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221
Martin Smith
2008-Jun-09 20:57 UTC
[asterisk-users] RFC2833 DTMF -- with an RTP debug log -- need someanalysis/interpretation
To add, here's one weird difference (how am I missing VLDTMF events?): Broken: sur-pbx-1:/home/martins# grep -i dtmf rfc2833-broken | grep -i chan_zap [Jun 9 16:26:21] DEBUG[11028] chan_zap.c: Started VLDTMF digit '2' [Jun 9 16:26:21] DEBUG[11028] chan_zap.c: Ending VLDTMF digit '2' Working: sur-pbx-1:/home/martins# grep -i dtmf rfc2833-working | grep -i chan_zap [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Started VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Ending VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Started VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Ending VLDTMF digit '2' [Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Started VLDTMF digit '1' [Jun 9 16:47:56] DEBUG[12300] chan_zap.c: Ending VLDTMF digit '1' Thanks :) Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of > Martin Smith > Sent: Monday, June 09, 2008 4:36 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] RFC2833 DTMF -- with an RTP debug > log -- need someanalysis/interpretation > > Hello all, > > I've got an Asterisk system I'm working on here, and we often dial > remote IVR systems, where our end must enter an extension to get to a > remote user. We're using Polycom hardphones here, speaking SIP, and > Asterisk sends these out over a PRI line with Zaptel hardware. > > I've used rtp debug on the phone, and I've got output, but I > can't tell > if it's correct or not -- I was dialing extension 221, but the remote > system lost one or more of the digits. I'd appreciate another > few pairs > of eyes checking out the rtp debug... > > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '2' received on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '2' on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on > SIP/199-b31ddc00, duration 60 ms > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end accepted with begin > '2' on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' has duration 60 > but want minimum 80, emulating on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end emulation of '2' > queued on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin '2' received on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin ignored '2' on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' received on > SIP/199-b31ddc00, duration 60 ms > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' has duration 60 > but want minimum 80, emulating on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end emulation of '2' > queued on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on > SIP/199-b31ddc00, duration 222 ms > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' put into dtmf > queue on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin emulation of '2' > with duration 100 queued on SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '1' received on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '1' on > SIP/199-b31ddc00 > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' received on > SIP/199-b31ddc00, duration 80 ms > [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' put into dtmf > queue on SIP/199-b31ddc00 > > Thanks! > > Martin Smith, Systems Developer > martins at bebr.ufl.edu > Bureau of Economic and Business Research > University of Florida > (352) 392-0171 Ext. 221 > > > _______________________________________________ > -- 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 >