Hello, I'm working on a (PJ)SIP trunking Asterisk machine with which I'm facing issues with DTMF. Installed version is 13.14.0.
1. In outbound calls SDP, I'm seeing these kind of lines: a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 I would expect events to range from 0 to 15, not to 16, as seen in rfc 4733 examples. What is this event 16 for ? Is there a way to configure this ? 2. With channel originated calls using Local prefix, I can read this on console: [2017-03-30 15:11:44] DTMF[11505][C-00000056]: channel.c:4103 __ast_read: DTMF begin '#' received on PJSIP/Foo-0000006f [2017-03-30 15:11:44] DTMF[11505][C-00000056]: channel.c:4114 __ast_read: DTMF begin passthrough '#' on PJSIP/Foo-0000006f [2017-03-30 15:11:44] DTMF[11501][C-00000057]: channel.c:4103 __ast_read: DTMF begin '#' received on Local/2@from-originate-00000024;1 [2017-03-30 15:11:44] DTMF[11501][C-00000057]: channel.c:4114 __ast_read: DTMF begin passthrough '#' on Local/2@from-originate-00000024;1 [2017-03-30 15:11:44] DTMF[11505][C-00000056]: channel.c:4017 __ast_read: DTMF end '#' received on PJSIP/Foo-0000006f, duration 180 ms With pure inbound-outbound calls (calls coming in from PJSIP and leaving through PJSIP), I get this: [2017-03-30 15:51:01] DTMF[11650][C-0000006d]: channel.c:4017 __ast_read: DTMF end '9' received on PJSIP/Bar-IPO-00000093, duration 100 ms [2017-03-30 15:51:01] DTMF[11650][C-0000006d]: channel.c:4044 __ast_read: DTMF begin emulation of '9' with duration 100 queued on PJSIP/Bar-IPO-00000093 [2017-03-30 15:51:01] DTMF[11650][C-0000006d]: channel.c:4181 __ast_read: DTMF end emulation of '9' queued on PJSIP/Bar-IPO-00000093 Can I get both inbound and outbound DTMF on console ? How ? 3. Looking at DTMF duration (as logged by Asterisk console), I can see that some (from mobile phone) have a 180ms duration while some, from an other SIP trunk, have a 100ms duration. Can I configure tone duration ? How ? Should I configure this ? Does this duration any real relation with the way a user presses keys ? Best regards
-- _____________________________________________________________________ -- 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
