I forgot to mention that pjsip.conf for this endpoint (that doesn't support telephone-event) already has this:
dtmf_mode=auto Cheers, Kingsley. On Tue, 2021-10-19 at 15:19 +0100, Kingsley Tart wrote: > Hi, > > I'm using Asterisk 18 to receive a call via SIP, dial a different SIP > destination and bridge them together. > > However, even if the destination indicates that it doesn't support > telephone-event, Asterisk is still sending DTMF as events, not > transcoding to inband. > > Asterisk is recognising inband DTMF coming in to it, but if it > receives > DTMF in RTP events it just forwards them on instead of transcoding. > > eg, the SDP in the INVITE that Asterisk sent out: > > v=0. > o=- 1051458170 1051458170 IN IP4 88.151.41.28. > s=Asterisk. > c=IN IP4 88.151.41.28. > t=0 0. > m=audio 13470 RTP/AVP 8 0 3 9 110 117 119 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:117 speex/16000. > a=rtpmap:119 speex/32000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=maxptime:60. > a=sendrecv. > > > The SDP in the response, notably without telephone-event: > > v=0. > s=sip call. > c=IN IP4 109.159.136.164. > t=0 0. > m=audio 63356 RTP/AVP 8 0 3 9. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=ptime:20. > a=sendrecv. > > Any idea how I can fix this? > > 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
