Re: [Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?

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Have to look at it later dude.
On May 23, 2013 11:36 AM, "Greg Horton" <gre...@gm...> wrote:

> Hi Chris,
>
> Here is output for core show channels as well as the more detailed "core
> show channel" for the source SIP channel I want to pass DTMF from...
>
>
> Channel              Location             State   Application(Data)
>
>
> SIP/1410-00000104    1024@from-internal:1 Up
> Playback(conf-invalidpin)
>
> 1 active channel
>
> 1 active call
>
> 193 calls processed
>
>
>
>
> -- General --
>
>            Name: SIP/1410-00000104
>
>            Type: SIP
>
>        UniqueID: 1369243057.298
>
>        LinkedID: 1369243057.298
>
>       Caller ID: 1410
>
>  Caller ID Name: GregGrandstream1410
>
> Connected Line ID: (N/A)
>
> Connected Line ID Name: (N/A)
>
>     DNID Digits: (N/A)
>
>        Language: en
>
>           State: Up (6)
>
>           Rings: 0
>
>   NativeFormats: 0x4 (ulaw)
>
>     WriteFormat: 0x4 (ulaw)
>
>      ReadFormat: 0x4 (ulaw)
>
>  WriteTranscode: No
>
>   ReadTranscode: No
>
> 1st File Descriptor: 25
>
>       Frames in: 2035
>
>      Frames out: 717
>
>  Time to Hangup: 0
>
>    Elapsed Time: 0h0m42s
>
>   Direct Bridge: <none>
>
> Indirect Bridge: <none>
>
>  --   PBX   --
>
>         Context: from-internal
>
>       Extension: 1024
>
>        Priority: 10
>
>      Call Group: 0
>
>    Pickup Group: 0
>
>     Application: Read
>
>            Data: PIN,enter-conf-pin-number,,,,
>
>     Blocking in: ast_waitfor_nandfds
>
>       Variables:
>
> READSTATUS=ERROR
>
> PLAYBACKSTATUS=SUCCESS
>
> PINCOUNT=2
>
> PIN=
>
> GOSUB_RETVAL=
>
> REC_STATUS=RECORDING
>
> MEETME_RECORDINGFORMAT=wav
>
>
> MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/2013/05/22/conf-1024-1024-20130522-131738-1369243057.298
>
> CALLFILENAME=conf-1024-1024-20130522-131738-1369243057.298
>
> DB_RESULT=conf-1024-1024-20130522-130848-1369242527.297
>
> FROMEXTEN=1410
>
> TIMESTR=20130522-131738
>
> YEAR=2013
>
> MONTH=05
>
> DAY=22
>
> NOW=1369243058
>
> REC_POLICY_MODE=always
>
> MON_FMT=wav
>
> MEETME_MUSIC=
>
> MAX_PARTICIPANTS=2
>
> MEETME_ROOMNUM=1024
>
> MACRO_DEPTH=0
>
> TTL=64
>
> CCSS_SETUP=TRUE
>
> AMPUSERCID=1410
>
> AMPUSERCIDNAME=GregGrandstream1410
>
> AMPUSER=1410
>
> REALCALLERIDNUM=1410
>
> SIPCALLID=454be043553cbd85499d0e5d688a50b3@54.235.67.221:5060
>
>
>   CDR Variables:
>
> level 1: recordingfile=conf-1024-1024-20130522-131738-1369243057.298.wav
>
> level 1: dnid=
>
> level 1: clid="GregGrandstream1410" <1410>
>
> level 1: src=1410
>
> level 1: dst=1024
>
> level 1: dcontext=from-internal
>
> level 1: channel=SIP/1410-00000104
>
> level 1: lastapp=Read
>
> level 1: lastdata=PIN,enter-conf-pin-number,,,,
>
> level 1: start=2013-05-22 13:17:37
>
> level 1: answer=2013-05-22 13:17:38
>
> level 1: duration=41
>
> level 1: billsec=40
>
> level 1: disposition=ANSWERED
>
> level 1: amaflags=DOCUMENTATION
>
> level 1: uniqueid=1369243057.298
>
> level 1: linkedid=1369243057.298
>
> level 1: sequence=363
>
>
> Thanks!
>
> Greg
>
>
> On Wed, May 22, 2013 at 9:11 PM, Chris Mylonas <ch...@mr...>wrote:
>
>> hi greg,
>> paste the output of freepbx's channel dump output.
>> what columns are available, you're looking at 'location' by the sounds of
>> it
>>
>> to expedite, i've doctored a snippet [1] for you from cli
>>
>> chris
>>
>>
>> [1]
>> from * cli it would look something like
>>
>> [cm@box5 ~]# asterisk -rx "core show channels"
>> Channel              Location/YourCustomDestination             State
>> Application(Data)
>> SIP/101-0000eb2d    1024@from-internal                          Up
>>  MeetMe(1024
>>
>> then use the SIP/101-xxxxxx channel
>>
>>
>>
>>
>>
>>
>> On Thu, May 23, 2013 at 2:01 AM, Greg Horton <gre...@gm...>wrote:
>>
>>> Hi Chris,
>>>
>>> When I look at the channel dump in FreePBX, it shows the destination as
>>> "1024@from-internal:1" while it's in the state of waiting for a PIN
>>> sequence.  So I tried using the A-J "sendAction(PlayDtmfAction)" interface
>>> with that destination, but no luck.  I get returnCode "Error".
>>>
>>> I also tried:
>>> 1024@from-internal
>>> Local/1024@from-internal
>>> Loca1/1024@from-internal:1
>>> Local/1024
>>>
>>> It seems like I am trying to pass the digit into the Dialplan at this
>>> point, and I am wondering if the sendAction(PlayDtmfAction) will not work
>>> for this case.  Maybe something else is required in A-J?
>>>
>>> Thanks,
>>> Greg
>>>
>>>
>>> On Tue, May 21, 2013 at 11:28 PM, Chris Mylonas <ch...@mr...>wrote:
>>>
>>>> When your dialer module is connected to the meetme room, issue a
>>>> dialplan function "core show channels" over AJ or on the asterisk CLI - you
>>>> might be able to see which channel your dialer module is using
>>>>
>>>> *shrug*
>>>>
>>>> HTH
>>>> Chris
>>>>
>>>>
>>>> On Wed, May 22, 2013 at 12:42 PM, GregLHorton <gre...@gm...>wrote:
>>>>
>>>>>
>>>>> Hi all,
>>>>>
>>>>> I have a client app with a dialer module, that can dial into a
>>>>> conference
>>>>> normally.  If my conference room is 1024, then I can pass 1024 to
>>>>> initiateCall() to dial into the conference.  If a password is
>>>>> required, the
>>>>> request for digits is played out by Asterisk.  At that time, digits
>>>>> entered
>>>>> in the dialer module do not get passed to the endpoint that detects
>>>>> the password digits.  This is because I normally use PlayDtmfAction
>>>>> for this
>>>>> purpose and that requires a channel like SIP/100-000034ab.  I do not
>>>>> see
>>>>> where this type of channel identifier is applicable to the conference
>>>>> room I
>>>>> just dialed into.  I have seen where a dummy channel ID relating to
>>>>> DAHDI
>>>>> shows up, but that is not until after I can successfully pass the PIN
>>>>> string.
>>>>>
>>>>> This is using MeetMe on Asterisk 10.0.
>>>>>
>>>>> I tried things like MEETME_ROOMNUM=1024, MEETME/1024, etc with no luck.
>>>>> Just wild guesses of course :).
>>>>>
>>>>> Any help appreciated!
>>>>>
>>>>> Greg
>>>>> --
>>>>> View this message in context:
>>>>> http://old.nabble.com/How-to-pass-DTMF-PIN-into-Meetme-using-A-J--tp35422999p35422999.html
>>>>> Sent from the Asterisk-Java Users mailing list archive at Nabble.com.
>>>>>
>>>>>
>>>>>
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>>>>>
>>>>
>>>>
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>>
>>
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>>
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>
>
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