[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.
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