[asterisk-users] Call recording - methodology

[Silver Thorne]

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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