Hi, I have built a new Asterisk installation:
root@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 <[email protected]> ? These are the files: kingsley@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. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
