[asterisk-bugs] [JIRA] (ASTERISK-25214) DTMF over SIP INFO and direct media does not work well together

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25214?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226716#comment-226716 ] 

Asterisk Team commented on ASTERISK-25214:
------------------------------------------

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].

> DTMF over SIP INFO and direct media does not work well together
> ---------------------------------------------------------------
>
>                 Key: ASTERISK-25214
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25214
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp, Channels/chan_sip/General
>    Affects Versions: 13.4.0
>            Reporter: Etienne Lessard
>         Attachments: annotated-sip-debug.txt, extensions.conf, sip.conf
>
>
> Hello,
> There seems to be some problems when using chan_sip with directmedia=yes and dtmfmode=info on asterisk 13.
> Given I have two users, Alice and Bob
> And Alice calls Bob (via Dial application, with no option)
> And Bob answers
> When Alice sends a *first* DTMF over SIP INFO
> Then Bob does not receive it (i.e. asterisk doesn't sent it)
> When Alice sends a *second* DTMF over SIP INFO
> Then Bob receives the *first* DTMF, with a duration equals to the time between the first and second DTMF that asterisk received
> What this means is that DTMF are not sent "promptly", the last DTMF is always missing, and the duration value of DTMF are unexpected.
> I've attached a sip.conf, extensions.conf and an annoted output of the asterisk CLI with "sip set debug on" and DTMF log displayed.
> As a note, this used to work correctly on asterisk 11.
> Thank you



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