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From: Yves A. <yv...@gm...> - 2013-06-03 15:00:39
|
hi,
just very short... i am in a hurry...
:1 means first leg of the call....
what you mean is jumping to a priority?? (11) you should be able to do
so... there is a parameter where you the specify the priority...
if thats not working... try to go with agi... let me know.... and
explain your objective... dialplan is not complete / fully clear 2 me...
yves
Am 03.06.2013 15:01, schrieb Greg Horton:
> Hi all,
>
> I am having a problem where the ext-meetme module in the dialplan gets
> to a point where it is waiting for PIN entry, after playing a
> recording to request that. This is the part of the Dialplan I refer to:
>
> [ 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]
>
> Is there an A-J command to specifically address 1024@ext-meetme, entry
> 11? I am trying to use PlayDtmfAction, where normally I can pass DTMF
> to a destination channel like SIP/101-000045c3a. Now, I have tried so
> many combinations and cannot get MeetMe to collect the digits of the PIN.
>
> When I make my call and collect Asterisk channel info, the destination
> channel at this time is seen as 1024@from-internal:1. Does the ":1"
> refer to entry 1 in the from-internal module of the dial plan?
>
> I have tried things like 1024@from-internal:1,
> Local/1024@from-internal:1, 1024@from-internal:11, and replacing
> from-internal with ext-meetme in all of these.
>
> If I cannot get directly to this point, is there some other roundabout
> way to achieve this goal?
>
> Please help as I am really stuck at this point.
>
> Thanks!
>
> Greg
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
|
|
From: Rounak S. <rou...@gm...> - 2013-06-03 14:47:25
|
Hello everyone, I have integrated my JAVA program(using asterisk-java) with a text-to-speech software where I send a string to the TTS software and it return me a .mp3 file (which can be converted to any desire format). Now I want this mp3 to be played to the caller. Which fastAGI command should I use to play the mp3? Do I have to first transfer my .mp3 file to Asterisk database and then ask it to run it?If yes,How should I transfer? |
|
From: Greg H. <gre...@gm...> - 2013-06-03 13:01:38
|
Hi all,
I am having a problem where the ext-meetme module in the dialplan gets to a
point where it is waiting for PIN entry, after playing a recording to
request that. This is the part of the Dialplan I refer to:
[ 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]
Is there an A-J command to specifically address 1024@ext-meetme, entry 11?
I am trying to use PlayDtmfAction, where normally I can pass DTMF to a
destination channel like SIP/101-000045c3a. Now, I have tried so many
combinations and cannot get MeetMe to collect the digits of the PIN.
When I make my call and collect Asterisk channel info, the destination
channel at this time is seen as 1024@from-internal:1. Does the ":1" refer
to entry 1 in the from-internal module of the dial plan?
I have tried things like 1024@from-internal:1, Local/1024@from-internal:1,
1024@from-internal:11, and replacing from-internal with ext-meetme in all
of these.
If I cannot get directly to this point, is there some other roundabout way
to achieve this goal?
Please help as I am really stuck at this point.
Thanks!
Greg
|
|
From: Atul A. <adi...@gm...> - 2013-06-02 17:10:51
|
Hii everyone, I am creating a IVR to listen music and I am using FastAGI protocol.I have the following two issues. First: After receiving a call,based on caller-ID,my Java program will query a soundFile database and plays a song.How can I do this? Please Note:My a*sterisk server does not have access to the database* and it * doesnot have the soundFile to play*.It is present with my Java server. As far as I know if I am using streamFileCommand the* filename should be present on my asterisk sound folder*.But here soundFile is with my javaServer. Second:I want to convert text to speech using festival(integrated with asterisk).Which command do I have to use? Please help -- Adi Agrawal |
|
From: Markus W. <mar...@ma...> - 2013-06-02 06:00:01
|
Hello Atul, First: be sure to use the latest version. There are tremendous differences between 0.3 and the last 1.X Versions even if they are milestones. Second: (That is my personal opinion). If something can be done via the dialplan you should do it there. Your project seems pretty easy to be done using the dialplan. Third: Thanks I don't need gmail. :-) Markus Am 02.06.2013 07:37, schrieb Atul Agrawal: > Hii, > I am new to asterisk-java so please bear with me. > > First:Which is the lastest stable version of asterisk-java? > I used the one in link below(0.3) but someone pointed out that this > one is not stable > http://sourceforge.net/projects/asterisk-java/ > I guess its 0.3.1 so I downloaded it from here(grepcode > <http://grepcode.com/snapshot/repo1.maven.org/maven2/org.asteriskjava/asterisk-java/0.3.1>) > > Second: I want to make a IVR where caller calls and information is > provided based on option selected by caller and the *caller phone > number*(no SIP).I will query the database and *create a voice > respone*. In the worst case *call will be directed to a human.* > > Can I do it using FastAGI only or do I have to use ManagerAPI or liveAPI? > > > -- > Atul Agrawal > B.E Hons. > BITS PILANI Goa Campus > Mobile:+91-81491-95049 > > > ------------------------------------------------------------------------------ > Get 100% visibility into Java/.NET code with AppDynamics Lite > It's a free troubleshooting tool designed for production > Get down to code-level detail for bottlenecks, with <2% overhead. > Download for free and get started troubleshooting in minutes. > http://p.sf.net/sfu/appdyn_d2d_ap2 > > > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
|
From: Atul A. <adi...@gm...> - 2013-06-02 05:37:29
|
Hii, I am new to asterisk-java so please bear with me. First:Which is the lastest stable version of asterisk-java? I used the one in link below(0.3) but someone pointed out that this one is not stable http://sourceforge.net/projects/asterisk-java/ I guess its 0.3.1 so I downloaded it from here(grepcode<http://grepcode.com/snapshot/repo1.maven.org/maven2/org.asteriskjava/asterisk-java/0.3.1> ) Second: I want to make a IVR where caller calls and information is provided based on option selected by caller and the *caller phone number*(no SIP).I will query the database and *create a voice respone*. In the worst case *call will be directed to a human.* Can I do it using FastAGI only or do I have to use ManagerAPI or liveAPI? -- Atul Agrawal B.E Hons. BITS PILANI Goa Campus Mobile:+91-81491-95049 |
|
From: Atul A. <adi...@gm...> - 2013-06-02 05:37:22
|
I've been using Gmail and thought you might like to try it out. Here's an invitation to create an account. You're Invited to Gmail! Atul Agrawal has invited you to open a Gmail account. Gmail is Google's free email service, built on the idea that email can be intuitive, efficient, and fun. Gmail has: *Less spam* Keep unwanted messages out of your inbox with Google's innovative technology. *Lots of space* Enough storage so that you'll never have to delete another message. *Built-in chat* Text or video chat with Atul Agrawal and other friends in real time. *Mobile access* Get your email anywhere with Gmail on your mobile phone. You can even import your contacts and email from Yahoo!, Hotmail, AOL, or any other web mail or POP accounts. Once you create your account, Atul Agrawal will be notified of your new Gmail address so you can stay in touch. Learn more<http://mail.google.com/mail/help/intl/en/about.html>or get started<http://mail.google.com/mail/a-3251cd0847-490dd1b35f-VsNPTejOcIegZGbp4T46uJXpfPI?pc=en-rf---a> ! Sign up<http://mail.google.com/mail/a-3251cd0847-490dd1b35f-VsNPTejOcIegZGbp4T46uJXpfPI?pc=en-rf---a> Google Inc. | 1600 Ampitheatre Parkway | Mountain View, California 94043 |
|
From: Miguel S. <m.s...@gm...> - 2013-06-01 06:11:29
|
Try creating an AMI connection inside the agi and using originateAction
instead.
El 31/05/2013 17:28, "Daniele Renda" <dan...@gm...> escribió:
> Hi,
> I need to make a call inside an agi script:
>
> In my
> public void service(AgiRequest request, AgiChannel channel) throws
> AgiException {
>
> I've this line:
> channel.exec("Dial", channel+ phoneNumber);
>
> But the call don't go out. I want to replace my dialplan in which now I've
> this:
>
> exten =>_XXX.,n,Dial(SIP/provider/${EXTEN})
>
> with this:
> exten =>_XXX.,n,Agi(agi://localhost/chiamatevoip.agi,ph003,${EXTEN})
>
> Thanks
>
> --
> Daniele Renda
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
|
|
From: Yves A. <yv...@gm...> - 2013-05-31 21:00:38
|
hi,
this smells like as if you´re in need of local channels...
pls. post the full context of dialplan thats used and the log from the
verbose cli and java
when the call is made.
solution will be quite easy, but requires further knowledge of the
mentioned details.
yves
Am 31.05.2013 17:27, schrieb Daniele Renda:
> Hi,
> I need to make a call inside an agi script:
>
> In my
> public void service(AgiRequest request, AgiChannel channel) throws
> AgiException {
>
> I've this line:
> channel.exec("Dial", channel+ phoneNumber);
>
> But the call don't go out. I want to replace my dialplan in which now
> I've this:
>
> exten =>_XXX.,n,Dial(SIP/provider/${EXTEN})
>
> with this:
> exten =>_XXX.,n,Agi(agi://localhost/chiamatevoip.agi,ph003,${EXTEN})
>
> Thanks
>
> --
> Daniele Renda
>
>
> ------------------------------------------------------------------------------
> Get 100% visibility into Java/.NET code with AppDynamics Lite
> It's a free troubleshooting tool designed for production
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
> http://p.sf.net/sfu/appdyn_d2d_ap2
>
>
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
|
|
From: Yves A. <yv...@gm...> - 2013-05-31 20:49:37
|
hi atul, 1.) the error says, that there is already an instance running on port 4573.. stop your instance before starting a new one... 2.) due to a bug(?) setting the port in fastagi-mapping.properties does not take effect... whats wrong with 4573? If you really need to change the port, you have to write a wrapper that instanciates DefaultAgiServer, then set the desired Port via setPort(int port) and than start the server... or take the sources and change the default port by yourself and recompile / pack the jar. 3.) you are using an outdated version of asterisk-java... regards, yves Am 31.05.2013 17:32, schrieb Atul Agrawal: > hii everyone, > I was trying FastAGI and I changed the port to 7856 as it was > showing this exception while running on 4573.Below is the error file. > > I added fastagi-mapping.properties in classpath in eclipse.I > attached the error file,fastagi-mapping.properties,and code below. > > > The link is a screenshot of errors and my > fastagi-mapping.properties.. > > Please Help > > https://docs.google.com/drawings/d/1_-VGjEJmIEAQrpXRk7GFBCubu155He4jC3pazSxu7E8/edit?usp=sharing > > -- > Atul Agrawal > B.E Hons. > BITS PILANI Goa Campus > Mobile:+91-81491-95049 > > > ------------------------------------------------------------------------------ > Get 100% visibility into Java/.NET code with AppDynamics Lite > It's a free troubleshooting tool designed for production > Get down to code-level detail for bottlenecks, with <2% overhead. > Download for free and get started troubleshooting in minutes. > http://p.sf.net/sfu/appdyn_d2d_ap2 > > > _______________________________________________ > 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
>>> 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: Atul A. <adi...@gm...> - 2013-05-31 15:32:46
|
hii everyone,
I was trying FastAGI and I changed the port to 7856 as it was showing
this exception while running on 4573.Below is the error file.
I added fastagi-mapping.properties in classpath in eclipse.I attached
the error file,fastagi-mapping.properties,and code below.
The link is a screenshot of errors and my fastagi-mapping.properties..
Please Help
https://docs.google.com/drawings/d/1_-VGjEJmIEAQrpXRk7GFBCubu155He4jC3pazSxu7E8/edit?usp=sharing
--
Atul Agrawal
B.E Hons.
BITS PILANI Goa Campus
Mobile:+91-81491-95049
|
|
From: Daniele R. <dan...@gm...> - 2013-05-31 15:27:24
|
Hi,
I need to make a call inside an agi script:
In my
public void service(AgiRequest request, AgiChannel channel) throws
AgiException {
I've this line:
channel.exec("Dial", channel+ phoneNumber);
But the call don't go out. I want to replace my dialplan in which now I've
this:
exten =>_XXX.,n,Dial(SIP/provider/${EXTEN})
with this:
exten =>_XXX.,n,Agi(agi://localhost/chiamatevoip.agi,ph003,${EXTEN})
Thanks
--
Daniele Renda
|
|
From: hiel m. <mat...@gm...> - 2013-05-30 19:14:47
|
hi all, I try to check Agent password before send a QueueAddAction. My question is very simple. How can I check agent password withAsterisk-java 1.0.0. CI and asterisk 11.2.1 Thanks a lot, Regards, Matthieu |
|
From: Yves A. <yv...@gm...> - 2013-05-27 19:57:14
|
the download is linked on the page you mentioned yourself... just look there again... yves Am 27.05.2013 18:44, schrieb Mohammad Daei: > As I can see in https://blogs.reucon.com/asterisk-java/ the latest > stable version is 0.3.1 > where can I download a newer version? > > > On Mon, May 27, 2013 at 9:07 PM, Mariusz Rutkowski <mk...@gm... > <mailto:mk...@gm...>> wrote: > > Hey Mohammad, > 0.3.1 is kind old version. Check if that errors occurs with latest > version >> Hi all >> I am using Asterisk-Java 0.3.1 with Asterisk 10 >> When I successfully connect to AMI I get a lot of of errors about >> not recognizing the events or setters like these: >> >> INFO: No event class registered for event type 'fullybooted', >> attributes: {status=Fully Booted, event=FullyBooted, >> privilege=system,all} >> >> >> May 27, 2013 8:38:58 PM >> org.asteriskjava.manager.internal.EventBuilderImpl setAttributes >> >> SEVERE: Unable to set property 'items' to '0' on >> org.asteriskjava.manager.event.StatusCompleteEvent: no setter >> >> >> there are a LOT of these error in my log. is this normal or am I >> doing something wrong? >> >> >> >> ------------------------------------------------------------------------------ >> 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... <mailto: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... > <mailto: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: Miguel S. <m.s...@gm...> - 2013-05-27 18:40:23
|
Yes, you can handle any kind of
event implementing and registering event listeners.
Take a look at the API documentation.
Anyway, you can filter the log setting your log4j.properties properly.
About your question asking where you can download it, I'm trying to
remember : github
Miguel
El 27/05/2013 20:06, "Mohammad Daei" <mo...@gm...> escribió:
> you mean there is a way to handle them?
> the problem is that it's really messing up the log (I'm using log4j).
> BTW Is there a newer version than 0.3.1. if there is, where can I download
> it?
>
>
> On Mon, May 27, 2013 at 9:57 PM, Miguel Santiago <
> m.s...@gm...> wrote:
>
>> This is normal if you don't handle this kind of events.
>> It should be a warning instead an error.
>> Don't worry if you don't have any appender to write these log to a
>> file/s..
>> El 27/05/2013 18:28, "Mohammad Daei" <mo...@gm...> escribió:
>>
>>> Hi all
>>> I am using Asterisk-Java 0.3.1 with Asterisk 10
>>> When I successfully connect to AMI I get a lot of of errors about not
>>> recognizing the events or setters like these:
>>>
>>> INFO: No event class registered for event type 'fullybooted',
>>> attributes: {status=Fully Booted, event=FullyBooted, privilege=system,all}
>>>
>>>
>>> May 27, 2013 8:38:58 PM
>>> org.asteriskjava.manager.internal.EventBuilderImpl setAttributes
>>>
>>> SEVERE: Unable to set property 'items' to '0' on
>>> org.asteriskjava.manager.event.StatusCompleteEvent: no setter
>>>
>>>
>>> there are a LOT of these error in my log. is this normal or am I doing
>>> something wrong?
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------
>>> 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: Mohammad D. <mo...@gm...> - 2013-05-27 18:05:15
|
you mean there is a way to handle them?
the problem is that it's really messing up the log (I'm using log4j).
BTW Is there a newer version than 0.3.1. if there is, where can I download
it?
On Mon, May 27, 2013 at 9:57 PM, Miguel Santiago
<m.s...@gm...>wrote:
> This is normal if you don't handle this kind of events.
> It should be a warning instead an error.
> Don't worry if you don't have any appender to write these log to a file/s..
> El 27/05/2013 18:28, "Mohammad Daei" <mo...@gm...> escribió:
>
>> Hi all
>> I am using Asterisk-Java 0.3.1 with Asterisk 10
>> When I successfully connect to AMI I get a lot of of errors about not
>> recognizing the events or setters like these:
>>
>> INFO: No event class registered for event type 'fullybooted', attributes:
>> {status=Fully Booted, event=FullyBooted, privilege=system,all}
>>
>>
>> May 27, 2013 8:38:58 PM
>> org.asteriskjava.manager.internal.EventBuilderImpl setAttributes
>>
>> SEVERE: Unable to set property 'items' to '0' on
>> org.asteriskjava.manager.event.StatusCompleteEvent: no setter
>>
>>
>> there are a LOT of these error in my log. is this normal or am I doing
>> something wrong?
>>
>>
>>
>> ------------------------------------------------------------------------------
>> 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: Miguel S. <m.s...@gm...> - 2013-05-27 17:27:14
|
This is normal if you don't handle this kind of events.
It should be a warning instead an error.
Don't worry if you don't have any appender to write these log to a file/s..
El 27/05/2013 18:28, "Mohammad Daei" <mo...@gm...> escribió:
> Hi all
> I am using Asterisk-Java 0.3.1 with Asterisk 10
> When I successfully connect to AMI I get a lot of of errors about not
> recognizing the events or setters like these:
>
> INFO: No event class registered for event type 'fullybooted', attributes:
> {status=Fully Booted, event=FullyBooted, privilege=system,all}
>
>
> May 27, 2013 8:38:58 PM org.asteriskjava.manager.internal.EventBuilderImpl
> setAttributes
>
> SEVERE: Unable to set property 'items' to '0' on
> org.asteriskjava.manager.event.StatusCompleteEvent: no setter
>
>
> there are a LOT of these error in my log. is this normal or am I doing
> something wrong?
>
>
>
> ------------------------------------------------------------------------------
> 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: Mohammad D. <mo...@gm...> - 2013-05-27 16:45:12
|
As I can see in https://blogs.reucon.com/asterisk-java/ the latest stable version is 0.3.1 where can I download a newer version? On Mon, May 27, 2013 at 9:07 PM, Mariusz Rutkowski <mk...@gm...>wrote: > Hey Mohammad, > 0.3.1 is kind old version. Check if that errors occurs with latest version > > Hi all > I am using Asterisk-Java 0.3.1 with Asterisk 10 > When I successfully connect to AMI I get a lot of of errors about not > recognizing the events or setters like these: > > INFO: No event class registered for event type 'fullybooted', > attributes: {status=Fully Booted, event=FullyBooted, privilege=system,all} > > > May 27, 2013 8:38:58 PM > org.asteriskjava.manager.internal.EventBuilderImpl setAttributes > > SEVERE: Unable to set property 'items' to '0' on > org.asteriskjava.manager.event.StatusCompleteEvent: no setter > > > there are a LOT of these error in my log. is this normal or am I doing > something wrong? > > > ------------------------------------------------------------------------------ > 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 lis...@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: Mariusz R. <mk...@gm...> - 2013-05-27 16:37:34
|
Hey Mohammad,
0.3.1 is kind old version. Check if that errors occurs with latest version
> Hi all
> I am using Asterisk-Java 0.3.1 with Asterisk 10
> When I successfully connect to AMI I get a lot of of errors about not
> recognizing the events or setters like these:
>
> INFO: No event class registered for event type 'fullybooted',
> attributes: {status=Fully Booted, event=FullyBooted, privilege=system,all}
>
>
> May 27, 2013 8:38:58 PM
> org.asteriskjava.manager.internal.EventBuilderImpl setAttributes
>
> SEVERE: Unable to set property 'items' to '0' on
> org.asteriskjava.manager.event.StatusCompleteEvent: no setter
>
>
> there are a LOT of these error in my log. is this normal or am I doing
> something wrong?
>
>
>
> ------------------------------------------------------------------------------
> 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: Yves A. <yv...@gm...> - 2013-05-27 16:35:29
|
please try again with latest release of AJ.
yves
Am 27.05.2013 18:26, schrieb Mohammad Daei:
> Hi all
> I am using Asterisk-Java 0.3.1 with Asterisk 10
> When I successfully connect to AMI I get a lot of of errors about not
> recognizing the events or setters like these:
>
> INFO: No event class registered for event type 'fullybooted',
> attributes: {status=Fully Booted, event=FullyBooted, privilege=system,all}
>
>
> May 27, 2013 8:38:58 PM
> org.asteriskjava.manager.internal.EventBuilderImpl setAttributes
>
> SEVERE: Unable to set property 'items' to '0' on
> org.asteriskjava.manager.event.StatusCompleteEvent: no setter
>
>
> there are a LOT of these error in my log. is this normal or am I doing
> something wrong?
>
>
>
> ------------------------------------------------------------------------------
> 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: Mohammad D. <mo...@gm...> - 2013-05-27 16:27:04
|
Hi all
I am using Asterisk-Java 0.3.1 with Asterisk 10
When I successfully connect to AMI I get a lot of of errors about not
recognizing the events or setters like these:
INFO: No event class registered for event type 'fullybooted', attributes:
{status=Fully Booted, event=FullyBooted, privilege=system,all}
May 27, 2013 8:38:58 PM org.asteriskjava.manager.internal.EventBuilderImpl
setAttributes
SEVERE: Unable to set property 'items' to '0' on
org.asteriskjava.manager.event.StatusCompleteEvent: no setter
there are a LOT of these error in my log. is this normal or am I doing
something wrong?
|
|
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: 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 01:33:57
|
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...
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