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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