[asterisk-users] DTMF emulation with SIP INFO and direct media
oza.4h07 at gmail.com
Fri Dec 15 05:12:32 CST 2017
1. Can you describe a bit further how both ends of the above call were both
made of and configured ?
DTMF receiving is Asterisk/SIP channel but which version ?
Is the other end a SIP phone or a SIP trunk ?
2. Do you observe such behaviour in a one-to-one setup (one end emits, the
other listen) or does the DTMF sending side also communicates with an other
2017-12-13 12:22 GMT+01:00 Jean Aunis <jean.aunis at prescom.fr>:
> 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[C-00000005] channel.c: DTMF end '#' received
> on SIP/xxx-00000004, duration 257 ms
> [Dec 13 11:56:16] DTMF[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[C-00000005] bridge_channel.c: Channel
> SIP/xxx-00000004 left 'native_rtp' basic-bridge <4a5905ac-29f8-41c5-9981-
> [Dec 13 11:56:19] DTMF[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.
> Jean Aunis
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