Hi, Trying to properly broadcast / relay DTMF digits to other confbridge users, but does not appear to work. Goal is to have a conference user be able to receive the DTMF, so it has the effect of being 'broadcasted.'
I have the following set up in 'confbridge.conf': dtmf_passthrough=yes >From logger.conf, I can see the DTMF tones via setting "console => dtmf". >When I dial into the conference bridge with a SIP UA and dial 9, for example, >this is what I see: sip1*CLI> [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4164 __ast_read: DTMF begin '9' received on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4175 __ast_read: DTMF begin passthrough '9' on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4078 __ast_read: DTMF end '9' received on SIP/3002-0000003d, duration 110 ms [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4119 __ast_read: DTMF end accepted with begin '9' on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4134 __ast_read: DTMF end '9' detected to have actual duration 59 on the wire, emulation will be triggered on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4141 __ast_read: DTMF end '9' has duration 59 but want minimum 80, emulating on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4198 __ast_read: DTMF end emulation of '9' queued on SIP/3002-0000003d sip1*CLI> So what is missing here or how to identify / troubleshoot? Is there an application that needs to pass the DTMF from the SIP user in sip.conf to the conference application? Thanks in advance, Jason -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
