[asterisk-users] DTMF
Jeff LaCoursiere
jeff at jeff.net
Fri Apr 17 15:28:26 CDT 2009
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
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