asterisk users - Oct 2021 - Asterisk 18 won't transcode DTMF to inband

On Wed, Oct 20, 2021 at 5:20 AM Kingsley Tart - Barritel Ltd <
kingsley.tart at barritel.com> wrote:
> On Tue, 2021-10-19 at 15:02 -0300, Joshua C. Colp wrote:
> > # asterisk -V
> > > Asterisk GIT-master-cc127a999cM
> > > #
> >
> > That's the master branch from around March or so, not 18.
>
> Wow, all this time I thought I was running 18! What version would it
> be? How can I tell?
>
It's not a version. It's the development branch at a point in time.

>
> Should I download and compile this instead?
>
>
>
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz

If you want to be running Asterisk 18 and a known released version, yes.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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On Wed, 2021-10-20 at 06:44 -0300, Joshua C. Colp wrote:
> > Should I download and compile this instead?
> > 
> >
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
> 
> If you want to be running Asterisk 18 and a known released version, yes. 
Right OK thanks, I'll do that and see whether it behaves any
differently.

Cheers,
Kingsley.

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

Cheers,
Kingsley.