Summary:	ASTERISK-03068: [request] dtmfmode=inband tone duration
Reporter:	ejayhire (ejayhire)	Labels:
Date Opened:	2004-12-22 08:15:22.000-0600	Date Closed:	2011-06-07 14:05:29
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/NewFeature
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	I would like to be able to specify the duration of the dtmf tones generated by dtmfmode=inband, preferably on a per peer basis. ****** ADDITIONAL INFORMATION ****** This is to solve a problem where too-short tones send to a non-asterisk device are not reliably recognized.  (a frequent problem when connecting ata's to pbx's.)
Comments:	By: Clod Patry (junky) 2004-12-22 09:36:27.000-0600 Make sure you assign the bug to the right project. By: Clod Patry (junky) 2005-01-01 01:46:03.000-0600 Can ya give us more details on it so we can work on this? I mean, in which case exactly it doesn't work? which hardwares? which settings? By: ejayhire (ejayhire) 2005-01-05 16:41:10.000-0600 Hello.  I will attempt to provide as much information as possible...  Sorry if this is long winded. I an using a cisco ATA188 with 3.0 code, and a recent stable version of asterisk.  Here is my Call Path. PRI -> zap card -> Asterisk server 1 -> Sip -> Asterisk server 2 -> Sip -> ATA -> analog phone. the sip.conf uses DtmfMode=Inband for everything. When we make an inbound (from the pstn to the pri, across the servers, to the ATA) call, and press a dtmf key on the PSTN phone, all that is heard on the ATA-side-analog phone is a short dtmf chirp, (I'd guess 100ms, but I can't measure it... it's too short), regardless of the duration of the pstn-phone-tone.   In our situation, we are connecting the fx ports on the ATA to a traditional pbx.  The pbx is too [dumb|slow|old|prissy] to reliably capture the short dtmf pulses.  Currently, the only way to increase the duration of a inband dtmf tone is to change a static value in rtp.c and recompile, and this affect everything on the pbx.  My feature request is to allow this to be set in the sip.conf on a per peer basis. As an aside, the ATA is supposed to support rfc2833 dtmf, but does not appear to work, and does not appear to support INFO at all. By: Mark Spencer (markster) 2005-01-24 08:00:16.000-0600 Are you sure you're changing something in rtp.c?  That would not be consistant with in-band DTMF. By: ejayhire (ejayhire) 2005-02-03 10:40:53.000-0600 I could be incorrect, and probably am because I can't find my notes... but I believe rtp.c is the code responsible for converting from dtmfmode=(anything that uses out of band signalling) into dtmfmode=inband. By: Olle Johansson (oej) 2005-02-13 13:22:57.000-0600 Closing this request until we have a patch. If you want result, add a bounty on the www.voip-info.org wiki. This bug report will be re-opened when we have a patch by the reporter or any bug marshal (available in #asterisk-bugs on the IRC channel). Thank you!