Hello Jean, 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 endpoint ? Cheers 2017-12-13 12:22 GMT+01:00 Jean Aunis <jean.au...@prescom.fr>: > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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