Hello,
Asterisk is said to handle call routing and codec translation.
I would like to force transcoding function with asterisk but when I try to
force transcoding I get the errors:
codec not compatible or
WARNING[4425]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't
make SIP/xxx compatible with SIP/yyy
How exactly works asterisk, in order to transcoding?
If you have any suggestions, hints, work around tricks I would appreciate them
much
Thanks in advance.
George
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