On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: > > > On Thu, 16 Apr 2009, Kevin P. Fleming wrote: > >> Jeff LaCoursiere wrote: >> >>> So may I assume that dtmfmode is inband only over IAX (since adding >>> compression seems to have killed it?). That would suck. >> >> No, DTMF is always out of band on IAX2, as long as Asterisk knows the >> DTMF is happening; if the DTMF is inband on the SIP channel, and >> Asterisk has been configured for non-inband DTMF on that channel, then >> it is not aware the DTMF is even present, so it just stays in the audio >> stream and gets compressed (and destroyed). >> >> You can verify this by adding the 'dtmf' logger channel to your console >> or a log file, and checking whether Asterisk is even aware of the DTMF >> events on the SIP channel. > > I went ahead and switched to SIP just for grins, and made sure > "dtmfmode=rfc2833" is in the peer config on both sides and in the entry > for the phone. So now it is: > > polycom501---SIP/ulaw--->ast1---SIP/g729--->ast2---IAX/ulaw--->ITSP > > looking at DTMF debug on ast2 I have: > > [Apr 17 15:18:06] DTMF[21585]: channel.c:2226 __ast_read: DTMF begin '5' > received on SIP/ahriise-0882f470 > [Apr 17 15:18:06] DTMF[21585]: channel.c:2236 __ast_read: DTMF begin > passthrough '5' on SIP/ahriise-0882f470 > [Apr 17 15:18:07] DTMF[21585]: channel.c:2148 __ast_read: DTMF end '5' > received on SIP/ahriise-0882f470, duration 200 ms > [Apr 17 15:18:07] DTMF[21585]: channel.c:2195 __ast_read: DTMF end > accepted with begin '5' on SIP/ahriise-0882f470 > [Apr 17 15:18:07] DTMF[21585]: channel.c:2211 __ast_read: DTMF end > passthrough '5' on SIP/ahriise-0882f470 > > Does this look like inband or out of band signaling?
Looking at it a little closer, some of the debug lines look different: [Apr 17 15:18:07] DTMF[7041]: channel.c:2226 __ast_read: DTMF begin '1' received on SIP/ahriise-0882f470 [Apr 17 15:18:07] DTMF[7041]: channel.c:2230 __ast_read: DTMF begin ignored '1' on SIP/ahriise-0882f470 [Apr 17 15:18:07] DTMF[21585]: channel.c:2148 __ast_read: DTMF end '1' received on SIP/ahriise-0882f470, duration 200 ms [Apr 17 15:18:07] DTMF[21585]: channel.c:2184 __ast_read: DTMF begin emulation of '1' with duration 200 queued on SIP/ahriise-0882f470 [Apr 17 15:18:07] DTMF[21585]: channel.c:2296 __ast_read: DTMF end emulation of '1' queued on SIP/ahriise-0882f470 Emulation? I am getting more confused by the moment. j > > I am starting to think the issue is actually at the ITSP, as I saw every > digit I pressed in the CLI on ast2, and yet the AT&T conference line I was > calling only recognized 3 out of six digits. > > Thanks, > > j > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
