Kingsley Tart
2021-Oct-19 14:19 UTC
[asterisk-users] Asterisk 18 won't transcode DTMF to inband
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.
Kingsley Tart
2021-Oct-19 14:46 UTC
[asterisk-users] Asterisk 18 won't transcode DTMF to inband
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. > >