Re: [Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?
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Re: [Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?
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From: Greg H. <gre...@gm...> - 2013-05-23 01:33:57
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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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>>>
>>> -- sent from web mail --
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> Try New Relic Now & We'll Send You this Cool Shirt
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>>> service
>>> that delivers powerful full stack analytics. Optimize and monitor your
>>> browser, app, & servers with just a few lines of code. Try New Relic
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>>
>>
>> ------------------------------------------------------------------------------
>> Try New Relic Now & We'll Send You this Cool Shirt
>> New Relic is the only SaaS-based application performance monitoring
>> service
>> that delivers powerful full stack analytics. Optimize and monitor your
>> browser, app, & servers with just a few lines of code. Try New Relic
>> and get this awesome Nerd Life shirt!
>> http://p.sf.net/sfu/newrelic_d2d_may
>> _______________________________________________
>> Asterisk-java-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>>
>>
>
>
> --
>
> -- sent from web mail --
>
>
> ------------------------------------------------------------------------------
> Try New Relic Now & We'll Send You this Cool Shirt
> New Relic is the only SaaS-based application performance monitoring service
> that delivers powerful full stack analytics. Optimize and monitor your
> browser, app, & servers with just a few lines of code. Try New Relic
> and get this awesome Nerd Life shirt! http://p.sf.net/sfu/newrelic_d2d_may
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
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