Thread: [Asterisk-java-users] How to pass DTMF PIN into Meetme using A-J?
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From: GregLHorton <gre...@gm...> - 2013-05-22 02:41:46
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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--tp35422995p35422995.html Sent from the Asterisk-Java Users mailing list archive at Nabble.com. |
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From: GregLHorton <gre...@gm...> - 2013-05-22 02:42:48
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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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From: Chris M. <ch...@mr...> - 2013-05-22 03:28:25
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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. > > > > ------------------------------------------------------------------------------ > 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 -- |
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From: Greg H. <gre...@gm...> - 2013-05-22 16:01:46
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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. >> >> >> >> ------------------------------------------------------------------------------ >> 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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From: Chris M. <ch...@mr...> - 2013-05-23 01:11:36
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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. >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> 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 >> >> > > > ------------------------------------------------------------------------------ > 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 -- |
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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.
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>> 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
>>>
>>>
>>
>>
>> ------------------------------------------------------------------------------
>> 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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From: Chris M. <ch...@mr...> - 2013-05-23 03:04:42
|
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. >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> 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 >>>> >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> 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 >> >> > > > ------------------------------------------------------------------------------ > 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 > > |
|
From: Greg H. <gre...@gm...> - 2013-05-23 03:18:51
|
No problem. I appreciate any time you have! On Wed, May 22, 2013 at 11:04 PM, Chris Mylonas <ch...@mr...> wrote: > 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. >>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> 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 >>>>> >>>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> 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 >>> >>> >> >> >> ------------------------------------------------------------------------------ >> 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 >> >> > > ------------------------------------------------------------------------------ > 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 > > |
|
From: Greg H. <gre...@gm...> - 2013-05-31 15:39:40
|
Bumping up as I have not solved this issue yet. This is the Dialplan area
for ext-meetme where I am getting stuck:
[ Context 'ext-meetme' created by 'pbx_config' ]
'1024' => 1. Macro(user-callerid,) [pbx_config]
2. Set(MEETME_ROOMNUM=1024) [pbx_config]
3. Set(MAX_PARTICIPANTS=2) [pbx_config]
4. Set(MEETME_MUSIC=${MOHCLASS}) [pbx_config]
5. Gosub(sub-record-check,s,1(conf,1024,always))
[pbx_config]
6. GotoIf($["${DIALSTATUS}" = "ANSWER"]?READPIN)
[pbx_config]
7. Answer() [pbx_config]
8. Wait(1) [pbx_config]
9. Set(PINCOUNT=0) [pbx_config]
[READPIN] 10. Read(PIN,enter-conf-pin-number,,,,) [pbx_config]
<THIS IS WHERE I WANT TO INJECT A DTMF STRING>
11. GotoIf($[x${PIN} = x123]?USER) [pbx_config]
12. GotoIf($[x${PIN} = x321]?ADMIN) [pbx_config]
13. Set(PINCOUNT=$[${PINCOUNT}+1]) [pbx_config]
14. GotoIf($[${PINCOUNT}>3]?h) [pbx_config]
15. Playback(conf-invalidpin) [pbx_config]
16. Goto(READPIN) [pbx_config]
[ADMIN] 17. Set(MEETME_OPTS=aAoTqcIMsr) [pbx_config]
18. Goto(STARTMEETME,1) [pbx_config]
[USER] 19. Set(MEETME_OPTS=oTqcIMsr) [pbx_config]
20. Goto(STARTMEETME,1) [pbx_config]
I have tried so many destination channel combos in PlayDtmfAction, using
1024@ preceded or not preceded by Local/, 1024@ext-meete or
1024@from-internal, then adding numbers at the end like ext-meetme:11 as I
thought that might force the DTMF into Dialplan entry 11.
Any other thoughts anybody?
Thanks,
Greg
On Wed, May 22, 2013 at 11:04 PM, Chris Mylonas <ch...@mr...> wrote:
> 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.
>>>>>>
>>>>>>
>>>>>>
>>>>>> ------------------------------------------------------------------------------
>>>>>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>> 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
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>>
>>
>> ------------------------------------------------------------------------------
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>> http://p.sf.net/sfu/newrelic_d2d_may
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>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
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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
>
>
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