| Summary: | ASTERISK-20975: DTMF issue with SIP trunk | ||
| Reporter: | Anna Vlasenko (foxy) | Labels: | |
| Date Opened: | 2013-01-23 08:17:07.000-0600 | Date Closed: | 2013-02-14 09:10:21.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | |
| Versions: | Frequency of Occurrence | Constant | |
| Related Issues: | |||
| Environment: | Cent OS 5.8 | Attachments: | |
| Description: | DTMF seems not work reliably for incoming calls through Skype Connect trunk.
The system is installed from FreePBX distro. Asterisk version is 1.8.8.0 There is a Skype connect trunk configured for LD calls. The problem is when users call a Skype number that leads to IVR, and try to reach any extension, Asterisk ignores DTMF tones. relaxdtmf=yes DTMF mode is RFC2833 (as Skype recommends) jitterbuffer is set to yes and that's what I see in logs. {noformat} [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '8' received on SIP/SkypeConnect-00000652 [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '8' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end '8' received on SIP/SkypeConnect-00000652, duration 240 ms [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end passthrough '8' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '9' received on SIP/SkypeConnect-00000652 [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '9' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '9' received on SIP/SkypeConnect-00000652, duration 240 ms [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '9' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin '4' received on SIP/SkypeConnect-00000652 [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin ignored '4' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '4' received on SIP/SkypeConnect-00000652, duration 240 ms [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '4' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin '3' received on SIP/SkypeConnect-00000652 [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin ignored '3' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end '3' received on SIP/SkypeConnect-00000652, duration 240 ms [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end passthrough '3' on SIP/SkypeConnect-00000652 {noformat} The duration seems to be ok. But what's about this DTMF ignoring? | ||
| Comments: | By: Michael L. Young (elguero) 2013-01-23 11:03:13.057-0600 Asterisk 1.8.8 is over a year old. There have been quite a few fixes put in since then for DTMF. Are you able to reproduce this on a newer version of Asterisk? If you do produce this on a newer version of Asterisk, can you provide what number is being dialed so we know what digits are being missed. Also, a full debug log would be helpful, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information. Thanks By: Rusty Newton (rnewton) 2013-02-14 09:10:13.939-0600 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||