[EMAIL PROTECTED] wrote:
Hi
I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by
outbound SIP.
Now i want to detect DTMF-Tone Code coming from the called party to
trigger a signal.
Can this be done with asterisk? I read that the codec with DTMF
detection are ulaw and alaw. But i couldn't find a command to detect
dtmf's within a normal call.
pbx-1*CLI> show application dial
pbx-1*CLI>
-= Info about application 'Dial' =-
[Synopsis]
Place a call and connect to the current channel
[Description]
Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):
This applicaiton will place calls to one or more specified channels. As soon
as one of the requested channels answers, the originating channel will be
answered, if it has not already been answered. These two channels will then
be active in a bridged call. All other channels that were requested will
then
be hung up.
Unless there is a timeout specified, the Dial application will wait
indefinitely until one of the called channels answers, the user hangs up, or
if all of the called channels are busy or unavailable. Dialplan
executing will
continue if no requested channels can be called, or if the timeout expires.
This application sets the following channel variables upon completion:
DIALEDTIME - This is the time from dialing a channel until when it
is disconnected.
ANSWEREDTIME - This is the amount of time for actual call.
DIALSTATUS - This is the status of the call:
CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER
| CANCEL
DONTCALL | TORTURE
For the Privacy and Screening Modes, the DIALSTATUS variable will be
set to
DONTCALL if the called party chooses to send the calling party to the
'Go Away'
script. The DIALSTATUS variable will be set to TORTURE if the called party
wants to send the caller to the 'torture' script.
This application will report normal termination if the originating
channel
hangs up, or if the call is bridged and either of the parties in the bridge
ends the call.
The optional URL will be sent to the called party if the channel
supports it.
If the OUTBOUND_GROUP variable is set, all peer channels created by this
application will be put into that group (as in Set(GROUP()=...).
Options:
A(x) - Play an announcement to the called party, using 'x' as the file.
C - Reset the CDR for this call.
d - Allow the calling user to dial a 1 digit extension while
waiting for
a call to be answered. Exit to that extension if it exists
in the
current context, or the context defined in the EXITCONTEXT
variable,
if it exists.
D([called][:calling]) - Send the specified DTMF strings *after* the
called
party has answered, but before the call gets bridged. The
'called'
DTMF string is sent to the called party, and the 'calling' DTMF
string is sent to the calling party. Both parameters can be used
alone.
f - Force the callerid of the *calling* channel to be set as the
extension associated with the channel using a dialplan 'hint'.
For example, some PSTNs do not allow CallerID to be set to
anything
other than the number assigned to the caller.
g - Proceed with dialplan execution at the current extension if the
destination channel hangs up.
G(context^exten^pri) - If the call is answered, transfer both
parties to
the specified priority. Optionally, an extension, or
extension and
context may be specified. Otherwise, the current extension
is used.
h - Allow the called party to hang up by sending the '*' DTMF digit.
H - Allow the calling party to hang up by hitting the '*' DTMF
digit.
j - Jump to priority n+101 if all of the requested channels were
busy.
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
variables can be used with this option:
* LIMIT_PLAYAUDIO_CALLER yes|no (default yes)
Play sounds to the caller.
* LIMIT_PLAYAUDIO_CALLEE yes|no
Play sounds to the callee.
* LIMIT_TIMEOUT_FILE File to play when time is up.
* LIMIT_CONNECT_FILE File to play when call begins.
* LIMIT_WARNING_FILE File to play as warning if 'y' is
defined.
The default is to say the time
remaining.
m([class]) - Provide hold music to the calling party until a requested
channel answers. A specific MusicOnHold class can be
specified.
M(x[^arg]) - Execute the Macro for the *called* channel before
connecting
to the calling channel. Arguments can be specified to the Macro
using '^' as a delimeter. The Macro can set the variable
MACRO_RESULT to specify the following actions after the Macro is
finished executing.
* ABORT Hangup both legs of the call.
* CONGESTION Behave as if line congestion was encountered.
* BUSY Behave as if a busy signal was encountered.
This will also
have the application jump to priority n+101
if the
'j' option is set.
* CONTINUE Hangup the called party and allow the calling
party
to continue dialplan execution at the next
priority.
* GOTO:<context>^<exten>^<priority> - Transfer the call to the
specified priority. Optionally, an extension, or
extension and priority can be specified.
n - This option is a modifier for the screen/privacy mode. It
specifies
that no introductions are to be saved in the priv-callerintros
directory.
N - This option is a modifier for the screen/privacy mode. It
specifies
that if callerID is present, do not screen the call.
o - Specify that the CallerID that was present on the *calling*
channel
be set as the CallerID on the *called* channel. This was the
behavior of Asterisk 1.0 and earlier.
p - This option enables screening mode. This is basically
Privacy mode
without memory.
P([x]) - Enable privacy mode. Use 'x' as the family/key in the
database if
it is provided. The current extension is used if a database
family/key is not specified.
r - Indicate ringing to the calling party. Pass no audio to the
calling
party until the called channel has answered.
S(x) - Hang up the call after 'x' seconds *after* the called party has
answered the call.
t - Allow the called party to transfer the calling party by
sending the
DTMF sequence defined in features.conf.
T - Allow the calling party to transfer the called party by
sending the
DTMF sequence defined in features.conf.
w - Allow the called party to enable recording of the call by
sending
the DTMF sequence defined for one-touch recording in
features.conf.
W - Allow the calling party to enable recording of the call by
sending
the DTMF sequence defined for one-touch recording in
features.conf.
pbx-1*CLI>
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