[asterisk-bugs] [JIRA] (ASTERISK-27848) RFC4733 DTMF Breaks When High Frequency Codecs are Offered

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27848?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243375#comment-243375 ] 

Asterisk Team commented on ASTERISK-27848:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> RFC4733 DTMF Breaks When High Frequency Codecs are Offered
> ----------------------------------------------------------
>
>                 Key: ASTERISK-27848
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27848
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 15.4.0
>         Environment: CentOS 7 based FreePBX/SNG 7 Distro running on a KVM based virtual machine
> Linphone Android and Linphone Windows softphones
>            Reporter: Dominic
>            Severity: Minor
>
> This is my first issue, so apologies if I didn't fill everything out quite right.
> I've been running into an issue where RFC4733 DTMF fails (the other side of the call doesn't recognize digits are pressed) when high frequency codecs are offered in the initial SIP INVITE.
> If I enable a codec in my Linphone settings that uses a frequency greater than 8000 Hz (e.g. speex, opus, etc.) even if that codec is not negotiated, DTMF fails.
> Apparently, Asterisk doesn't support telephone-event/16000, but simply having that codec present in the INVITE SDP shouldn't prevent telephone-event/8000 from working.
> More details and server logs are available here: https://community.asterisk.org/t/help-debugging-rfc4733-dtmf/74602



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