[asterisk-bugs] [JIRA] (ASTERISK-29051) bridge_simple: Asterisk should not enter native bridge if dtmf modes differ

Asterisk Team (JIRA) noreply at issues.asterisk.org
Fri Aug 28 04:32:43 CDT 2020


    [ https://issues.asterisk.org/jira/browse/ASTERISK-29051?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=251823#comment-251823 ] 

Asterisk Team commented on ASTERISK-29051:
------------------------------------------

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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

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> bridge_simple: Asterisk should not enter native bridge if dtmf modes differ
> ---------------------------------------------------------------------------
>
>                 Key: ASTERISK-29051
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29051
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_native_rtp, Bridges/bridge_simple, Resources/res_pjsip_sdp_rtp
>    Affects Versions: 17.6.0
>         Environment: Debian Buster
>            Reporter: Sebastian Damm
>            Severity: Minor
>
> When bridging two call legs where one leg supports rfc4733 events and the other leg does not, DTMF tones don't get converted. This is because Asterisk enters native bridge if codecs are equal and then has no chance to detect anything inside the rtp stream. When transcoding from one codec to another, Asterisk stays in simple bridge, and it should behave the same way if dtmf modes differ. 
> To reproduce: Set dtmf_mode to "auto" in the endpoint settings in pjsip.conf. Send a call from a client only supporting inband DTMF to the Asterisk, send this call to another client supporting telephone-events. Then send DTMF digits from the calling device. They will end up inband on the receiving client. However, if the receiving client is for example another Asterisk, it will not look into the audio if the SDP offered telephone-event. DTMF digits will not be recognized.



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