Jean Aunis
2017-Dec-13 11:22 UTC
[asterisk-users] DTMF emulation with SIP INFO and direct media
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171213/2fe9310a/attachment.html>
Olivier
2017-Dec-15 11:12 UTC
[asterisk-users] DTMF emulation with SIP INFO and direct media
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.aunis at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171215/d6cc7bf8/attachment.html>
Jean Aunis
2017-Dec-15 13:44 UTC
[asterisk-users] DTMF emulation with SIP INFO and direct media
Asterisk is in version 14.7.1. One end is a SIP Trunk to another Asterisk, the other end a home-made SIP phone. SIP INFO requests are coming from the other Asterisk. Both endpoints use chan_sip with "dtmfmode" set to "info". This is not strictly speaking a one-to-one setup since we're connecting to a SIP Trunk which then connects to another SIP phone, but I think it doesn't make much difference regarding SIP INFO handling. Le 15/12/2017 ? 12:12, Olivier a ?crit?:> 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.aunis at prescom.fr > <mailto:jean.aunis at 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/ <https://community.asterisk.org/> > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > <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 > <http://lists.digium.com/mailman/listinfo/asterisk-users> > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171215/3ff21dee/attachment.html>