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

[Kingsley Tart (JIRA)]

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

Kingsley Tart updated ASTERISK-29700:
-------------------------------------

    Description: 
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

  was:
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.



> 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
>         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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