Hi Dan et al;
I had actually done a sip reload, dialplan reload, module reload
res_features.so and logger reload.
However, upon seeing your email, I restarted the Asterisk server
completely to see if I had missed anything. I still see the same behaviour.
I am at a loss.
Glen
On 4/10/2011 14:37, Dan Journo wrote:
> I set the logger.conf to show reading of DTMF tones as per your
instructions below. This is what I see:
> [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin '*' received on
SIP/6000-0000002e
> [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin passthrough '*'
on SIP/6000-0000002e
> [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end '*' received on
SIP/6000-0000002e, duration 186 ms
> [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end accepted with begin
'*' on SIP/6000-0000002e
> [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end passthrough '*' on
SIP/6000-0000002e
> [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin '1' received on
SIP/6000-0000002e
> [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin passthrough '1'
on SIP/6000-0000002e
> [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end '1' received on
SIP/6000-0000002e, duration 193 ms
> [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end accepted with begin
'1' on SIP/6000-0000002e
> [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end passthrough '1' on
SIP/6000-0000002e
It looks like Asterisk hasnt added the new details from features.conf.
You may need to fully restart Asterisk in order to get this to work.
Dan Journo
Kesher Communications (UK)
Business Phone Systems <http://www.keshercommunications.com/> | Hosted
PBX <http://www.keshercommunications.com/hostedpbx.html>
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