Imran Ahmed wrote:
may or may not work, try at your own risk:
1) Use a sip soft phone and set the dtmf mode = inband.
2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or
info. (this is done so that asterisk ignores the inband dtmf on the
sip channel).
3) Design your dialplan such that asterisk should not depend on dtmf
from the sip call.
ex:
exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room
exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room.
exten xxx, 3, meetme(conference room)
Thank you very much.
I tried sjphone setting clinet and asterisk as above and it seems to
work. I will test it better in the next hours.
I had a look at meetme.c and i found a portion of code that manage dtmf
if ((f->frametype == AST_FRAME_DTMF) && (confflags &
CONFFLAG_EXIT_CONTEXT)) {
......
......
-----
I think this part manage the case of meetme application is called with
p, X or s option,
but maybe also (i'm not sure, i had not the time to study well enough
the source, and over all i'm not a so good c programmer)
that this part of code prevents asterisk to broadcast the sound to other
channels when it is not inband.
Sorry if my bad english make me not very clear.
Anyway, thank you very much to all for your help.
Accursio Avona
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