[asterisk-bugs] [JIRA] (ASTERISK-20975) DTMF issue with SIP trunk
Michael L. Young (JIRA)
noreply at issues.asterisk.org
Wed Jan 23 11:04:58 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-20975?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=202138#comment-202138 ]
Michael L. Young commented on ASTERISK-20975:
---------------------------------------------
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
> DTMF issue with SIP trunk
> -------------------------
>
> Key: ASTERISK-20975
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20975
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Environment: Cent OS 5.8
> Reporter: Anna Vlasenko
>
> 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?
>
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