[asterisk-users] Asterisk 18 won't transcode DTMF to inband

[Joshua C. Colp]

On Fri, Oct 22, 2021 at 11:07 AM Kingsley Tart <kingsley at dns99.co.uk> wrote:

> Hi,
>
> I have built a new Asterisk installation:
>
> root at gw9:/tmp# asterisk -V
> Asterisk 18.7.1
>
> It still does the same thing, which is
>
> a. Asterisk receives INVITE containing SDP telephone-event
> b. Asterisk uses Dial with pjsip and sends INVITE to destination
> including SDP telehone-event
> c. Asterisk receives 200 OK back from destination WITHOUT telephone-
> event
> d. Asterisk forwards DTMF received to the destination in RTP events
>
> I've grabbed some debug info as per
> https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
>  and also have a pcap file containing all SIP and RTP.
>
> To save me spamming list list, may I send these files to your personal
> email address Joshua C. Colp <jcolp at sangoma.com> ?
>
> These are the files:
>
> kingsley at gandalf:/tmp$ ls -l *gz
> -rw-r--r-- 1 kingsley kingsley  40813 Oct 22 15:00 astlog.gz
> -rw-rw-r-- 1 kingsley kingsley 358895 Oct 22 14:57 dtmf-test.pcap.gz
>
> pjsip.conf contains these settings 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
>

I don't provide direct support like that. As there seems to be a bug and
you have a case that reproduces it with logs, then you can file an issue[1]
and the current individual doing bug triage will look. If it is accepted
there is no time frame on when it would get looked into and resolved.

[1] https://issues.asterisk.org/jira

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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