[asterisk-bugs] [JIRA] (ASTERISK-27848) rtp: DTMF Breaks With telephony-event/16000

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27848?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-27848:
-----------------------------------

    Assignee: Dominic
      Status: Waiting for Feedback  (was: Triage)

Can you please attach all the information here instead of linking to the forum? For those who work on issues it makes it easier to keep everything together, and also if people in the future want to reference back.

> rtp: DTMF Breaks With telephony-event/16000
> -------------------------------------------
>
>                 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
>            Assignee: 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



--
This message was sent by Atlassian JIRA
(v6.2#6252)