[asterisk-bugs] [JIRA] (ASTERISK-29700) DTMF is sent in RTP events even if remote 200 OK omits telephone-event

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29700?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-29700:
-------------------------------------

    Assignee: Asterisk Team  (was: Kingsley Tart)
      Status: Triage  (was: Waiting for Feedback)

> DTMF is sent in RTP events even if remote 200 OK omits telephone-event
> ----------------------------------------------------------------------
>
>                 Key: ASTERISK-29700
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29700
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 18.7.1
>         Environment: Debian 10. Asterisk compiled from source.
>            Reporter: Kingsley Tart
>            Assignee: Asterisk Team
>         Attachments: astlog.gz, dtmf-test.pcap.gz
>
>
> a. Asterisk receives INVITE containing SDP telephone-event
> b. Asterisk uses Dial with pjsip and sends INVITE to destination
> including SDP telephone-event
> c. Asterisk receives 200 OK back from destination WITHOUT telephone-
> event
> d. Asterisk forwards DTMF received to the destination in RTP events instead of falling back to inband audio
> We do have spandsp installed (if that's relevant?). This Asterisk installation DOES recognise inband audio DTMF being sent in to it.
> pjsip.conf has the following for the destination endpoint:
> [opensips-ipx]
> type=endpoint
> send_rpid=no
> trust_id_inbound=yes
> ; change this when we write the custom context for it:
> context=from-pubopensips
> aors=opensips-ipx-vip-a,opensips-ipx-vip-b,opensips-ipx-vip-c
> redirect_method=uri_pjsip
> disallow=all
> allow=alaw
> allow=ulaw
> allow=g722
> dtmf_mode=auto



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