Kingsley Tart
2021-Oct-22 14:07 UTC
[asterisk-users] Asterisk 18 won't transcode DTMF to inband
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.
Joshua C. Colp
2021-Oct-22 14:11 UTC
[asterisk-users] Asterisk 18 won't transcode DTMF to inband
On Fri, Oct 22, 2021 at 11:07 AM Kingsley Tart <kingsley at dns99.co.uk> wrote:> 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 >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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20211022/f00e99e7/attachment.html>