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

Kingsley Tart kingsley at dns99.co.uk
Fri Oct 22 09:07:12 CDT 2021


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.




More information about the asterisk-users mailing list