[asterisk-bugs] [JIRA] (ASTERISK-29700) DTMF is sent in RTP events even if remote 200 OK omits telephone-event
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Mon Oct 25 09:39:49 CDT 2021
[ 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
> Severity: Trivial
> Labels: fax, webrtc
> 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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