[asterisk-users] DTMF emulation with SIP INFO and direct media

[Jean Aunis]

Hello,

I think there is an issue when DTMF are handled with SIP INFO and direct 
media is enabled.

When I receive a SIP INFO, the logs tell me that a "DTMF begin" is 
generated, but no related "DTMF end" is generated, unless the call is 
ended. Here is an excerpt of the logs :

*--- SIP INFO received **on **SIP/xxx-00000004:*

[Dec 13 11:56:16] DTMF[18193][C-00000005] channel.c: DTMF end '#' 
received on SIP/xxx-00000004, duration 257 ms
[Dec 13 11:56:16] DTMF[18193][C-00000005] channel.c: DTMF begin 
emulation of '#' with duration 257 queued on SIP/xxx-00000004

*--- **SIP/xxx-00000004 **is hanged up:*

[Dec 13 11:56:19] VERBOSE[18193][C-00000005] bridge_channel.c: Channel 
SIP/xxx-00000004 left 'native_rtp' basic-bridge 
<4a5905ac-29f8-41c5-9981-e9d0f4966c56>
[Dec 13 11:56:19] DTMF[18193][C-00000005] bridge_channel.c: DTMF end '#' 
simulated to bridge 4a5905ac-29f8-41c5-9981-e9d0f4966c56 because 
SIP/xxx-00000004 left.  Duration 3012 ms.

Do you think it is a bug ? I would tend to say yes, but I'm not so sure.

Regards

Jean Aunis

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