Summary:	ASTERISK-00168: DTMF tones on remote IVRs are to short to be detected
Reporter:	yamez (yamez)	Labels:
Date Opened:	2003-08-25 14:03:06	Date Closed:	2011-06-07 14:05:04
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	DTMF tones on remote IVRs are to short to be detected I get this with info,rfc2833 and inband. ****** ADDITIONAL INFORMATION ****** Could be related to bug_id=0000130?
Comments:	By: brentonr (brentonr) 2003-08-25 15:32:10 I'm using an AudioCodes MP-104 FXO box, with snom 200 handsets: snom200 ---> asterisk ---> MP104FXO ---> POTS When using rfc2833 or inband (using G.711u/a), either way (verified with 'sip show channel ...'), the DTMF tones come across as very short - and in the case of '1' and '9', they are very screechy. I'm using CVS-08/14/03-09:03:21, also found this problem in CVS-07/29/03-09:40:44. By: Mark Spencer (markster) 2003-08-25 17:36:23 The value should be plenty long already (assuming they're really being inserted at all).  If you change the value in digits.h and make it longer does it improve? By: brentonr (brentonr) 2003-08-26 08:29:58 digits.h is part of the zaptel tree, isn't it?  Does that make a difference in my situation? By: yamez (yamez) 2003-08-26 08:33:05 Changeing digits.h to #define DEFAULT_DTMF_LENGTH     200 * 8 #define DEFAULT_MFV1_LENGTH     120 * 8 did nothing. By: yamez (yamez) 2003-08-26 09:00:50 This bug report should be under SIP (I thought that was were I put it) If you put sip.conf: dtmfmode=info in [general]  outgoing tones are as long as the key press. Works fine with any IVR but incoming tones don't work for me inside Asterisk.   I think this is because I don't know how to make my cisco AS5300 send info instead of (rfc2833 or inband). So I can't use this work around. But if your pots gatway can send info, let me know if that works for you.   I not sure what dtmfmode in [general] does. <smile> I'm guessing that it changes how Asterisk sends tones when not talking to a registered phone? By: Mark Spencer (markster) 2003-08-26 09:17:03 Ahhh...  So Asterisk is not the thing actually terminating the tone?  The MP104FXO is? By: brentonr (brentonr) 2003-08-26 10:20:16 I'm not sure what's doing what anymore, perhaps I'm just lost :) I'm using SIP from the snom 200 to asterisk, and bridging that to the MP104FXO.  I've tried 'info', 'rfc2833', and 'inband' options (set on the snom) for connecting to asterisk, and can't get DTMF to correctly come across on the remote POTS end (originating from the MP104FXO). Note that I don't think it's the snom 200's fault; I can get voicemail and other DTMF to work fine when it's just snom 200 ---> asterisk. By: yamez (yamez) 2003-08-26 10:57:31 Asterisk terminatins a voip call.   In my case: cisco7960  -sip--> Asterisk --sip--> AS5300 ---> pots or cisco7960 -sip--> Asterisk(box 1) --sip--> Asterisk(box 2) ----> pots or cisco7960 -sip--> Asterisk --sip--> vocal --sip--> pots I have diffrent gateways in diffrent citys. The tones used to be the right length, then an upgrade of Asterisk broke tones via rfc2833. By: yamez (yamez) 2003-08-26 10:58:51 The error is some problem with sip to sip bridging. By: yamez (yamez) 2003-08-27 10:06:02 I have a work-around for this bug. I have set all my phones to dtmfmode=rfc2833 and all my gateway peers to dtmfmode=info. I have no dtmfmode in the [general] section. This work-around work as long as your gatway can ecxept info and send rfc2833 both Vocal and my Ciscos can. By: brentonr (brentonr) 2003-08-27 11:54:47 That workaround also works for me with the AudioCodes MP104 FXO. *whew :)* By: John Todd (jtodd) 2003-09-08 18:27:25 Marking this as resolved, since it seems that this was just a usage issue.