Hello List,
I have seen that when ever asterisk gets a SIP INFO request from a SIP channel
it generates the requested DTMF tone and writes to the destination channel also
it forwards the SIP INFO message. As I am very new to this domain, it is really
confusing me. Why not asterisk writes only
Hello List,
We are using 'F' parameter in meetme Dialplan application to broadcast SIP INFO
(1 and 0) as DTMF tone to all the participants.
The DTMF configuration for all the connected SIP clients is SIP INFO.
The problem we are seeing, asterisk is taking some time to broadcast the SIP
INFO
