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? 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