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
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
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
