[asterisk-users] RFC2833 DTMF -- with an RTP debug log -- need someanalysis/interpretation

[Martin Smith]

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 
> 
> 
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