On Fri, Oct 22, 2021 at 11:07 AM Kingsley Tart <[email protected]> wrote:
> 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 > 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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