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 :
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
<mailto: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
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