[asterisk-users] Improvement of PJSIP dtmf_mode description

[Olivier]

Hello,

While debugging a SIP trunk with an Avaya IPO, I noticed that wiki's PJSIP
dtmf_mode at [1] includes:

"This setting allows to choose the DTMF mode for endpoint communication.

    rfc4733 - DTMF is sent out of band of the main audio stream. This
supercedes the older RFC-2833 used within the older chan_sip.
    inband - DTMF is sent as part of audio stream.
    info - DTMF is sent as SIP INFO packets.
    auto - DTMF is sent as RFC 4733 if the other side supports it or as
INBAND if not.
    auto_info - DTMF is sent as RFC 4733 if the other side supports it or
as SIP INFO if not."


The above description doesn't mention anything about incoming DTMF
treatment.
May I suggest that:
- either dtmf_mode has no influence itself on incoming DTMF treatment and
it could be explicitely mentioned,
- either dtmf_mode has an influence itself on incoming DTMF treatment and
this could be described.

What do you think of this ?

Best regards

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_dtmf_mode
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