DTMF Emulation With SIP INFO And Direct Media

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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
[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.


Jean Aunis

2 thoughts on - 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 ?


    2017-12-13 12:22 GMT+01:00 Jean Aunis :

  • 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 à 12:12, Olivier a écrit :