[asterisk-users] Call recording - methodology

[Dan Journo]

> 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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