| Summary: | ASTERISK-06179: rfc2833 dtmf is unrecognizable with 1.2.1 | ||
| Reporter: | Anthony Dean (tallenglish) | Labels: | |
| Date Opened: | 2006-01-24 08:04:43.000-0600 | Date Closed: | 2006-01-24 09:09:16.000-0600 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | we are using asterisk stable 1.2.1 (tgz download, not from cvs). and we are having issues with dtmf tones when using rfc2833. we dont have any jitter as far as I know as we can't hear any choppy speech and using inband dtmf clears the issue for the most part. rfc2833 is what all our carriers use and was working in 1.0.9 (and the 1.2.0rc1) if you listen to the tones they sound very short and almost like a s.o.s. dot instead of a s.o.s. dash (sometimes we don't here the tones at all) so I am guessing the sound is too short to be recognized by most systems. we have tried asterisk -> openser -> carrier, asterisk -> sysmaster -> carrier and asterisk -> as5350 (sip -> tdm) -> carrier all which produce the same results (openser and sysmaster may have issues but asterisk should work flawlessly with the cisco as5350. as a test I used a linksys PAP2-NA using the same sip paths described above (and it is set to use rfc2833 by default) and tones come through perfectly). we are using cisco 7940 phones to origionate the calls with asterisk. when we use inband we get lots of the following errors which flood the messages file pretty badly (1000+ messages every second while the call is active, increasing our usual log file from 20+ MB to close to 160MB) Jan 24 09:42:17 WARNING[31075] dsp.c: Inband DTMF is not supported on codec g729. Use RFC2833 would it not be better to just send one warning per call id? if you want any debug output please let me know what you want. | ||
| Comments: | By: Anthony Dean (tallenglish) 2006-01-24 08:12:04.000-0600 rfc2833 dtmf from the phone to asterisk (as used in voicemail for example) works perfectly fine - in fact inband doesn't work at all between the phone and asterisk (at least when testing voicemail between the cisco 7940's and asterisk). By: Anthony Dean (tallenglish) 2006-01-24 08:57:33.000-0600 I tried 1.2.1, 1.2.2 (both with issues with rfc2833 dtmf). I rolled back to 1.0.9 and dtmf issues were resolved (could use rfc2833 as normal). for now I am going to stick with 1.0.9 but whatever was changed with rfc2833 dtmf (specifically sip calls, haven't tested to see if iax is also effected) doesn't work with nextone, as5350 or openser (although I don't think the last one does anything with dtmf as its a pure sip router and doesn't do anything with rtp). | ||