Не работает Playback и Background...Помогите плиз
Решил сделать голосовую штуку.....
Но вот что мне пишет Астериск
[Oct 31 02:37:22] WARNING[1505]: pbx.c:4047 pbx_extension_helper: No application 'Playback ' for extension (6637202_in, 00044404, 4)
Покурил вот это:
http://www.asteriskguru.com/tutorials/di ... tions.html
и вот это
http://www.asteriskguru.com/tutorials/no ... nsion.html
Апликейшены не могу посмотреть, потому что у меня даже нет команды
show applications
Что я делаю не так: Подскажите, пожалуйста, чайеику...
Вот что я пытался сделать
[6637202_in]
exten => 00044404,1,Answer
exten => 00044404,n,Set(CALLERID(num)=8${CALLERID(num)})
exten => 00044404,n,Dial(SIP/20&SIP/21,30,m)
exten => 00044404,n,Playback (good-by)
exten => 00044404,n,Hangup
| safronov_alex wrote: |
| Что я делаю не так: Подскажите, пожалуйста, чайеику... |
core show help
| safronov_alex wrote: |
| Апликейшены не могу посмотреть, потому что у меня даже нет команды show applications |
| Code: |
| *CLI> core show version Asterisk 1.8.0 built by root @ desktop on a x86_64 running Linux on 2010-10-27 03:40:40 UTC *CLI> core show applications -= Registered Asterisk Applications =- AddQueueMember: Dynamically adds queue members. ADSIProg: Load Asterisk ADSI Scripts into phone AgentLogin: Call agent login. AgentMonitorOutgoing: Record agent's outgoing call. AGI: Executes an AGI compliant application. AlarmReceiver: Provide support for receiving alarm reports from a burglar or fire alarm panel. AMD: Attempt to detect answering machines. Answer: Answer a channel if ringing. Authenticate: Authenticate a user BackGround: Play an audio file while waiting for digits of an extension to go to. BackgroundDetect: Background a file with talk detect. Bridge: Bridge two channels. Busy: Indicate the Busy condition. CallCompletionCancel: Cancel call completion service CallCompletionRequest: Request call completion service for previous call CELGenUserEvent: Generates a CEL User Defined Event. ChangeMonitor: Change monitoring filename of a channel. ChanIsAvail: Check channel availability ChannelRedirect: Redirects given channel to a dialplan target ChanSpy: Listen to a channel, and optionally whisper into it. ClearHash: Clear the keys from a specified hashname. ConfBridge: Conference bridge application. Congestion: Indicate the Congestion condition. ContinueWhile: Restart a While loop. ControlPlayback: Play a file with fast forward and rewind. DAHDIBarge: Barge in (monitor) DAHDI channel. DAHDIRAS: Executes DAHDI ISDN RAS application. DAHDIScan: Scan DAHDI channels to monitor calls. DAHDISendCallreroutingFacility: Send QSIG call rerouting facility over a PRI. DAHDISendKeypadFacility: Send digits out of band over a PRI. DateTime: Says a specified time in a custom format. DBdel: Delete a key from the asterisk database. DBdeltree: Delete a family or keytree from the asterisk database. DeadAGI: Executes AGI on a hungup channel. Dial: Attempt to connect to another device or endpoint and bridge the call. Dictate: Virtual Dictation Machine. Directory: Provide directory of voicemail extensions. DISA: Direct Inward System Access. DumpChan: Dump Info About The Calling Channel. EAGI: Executes an EAGI compliant application. Echo: Echo audio, video, DTMF back to the calling party EndWhile: End a while loop. Exec: Executes dialplan application. ExecIf: Executes dialplan application, conditionally. ExecIfTime: Conditional application execution based on the current time. ExitWhile: End a While loop. ExtenSpy: Listen to a channel, and optionally whisper into it. ExternalIVR: Interfaces with an external IVR application. Festival: Say text to the user. Flash: Flashes a DAHDI Trunk. FollowMe: Find-Me/Follow-Me application. ForkCDR: Forks the Call Data Record. GetCPEID: Get ADSI CPE ID. Gosub: Jump to label, saving return address. GosubIf: Conditionally jump to label, saving return address. Goto: Jump to a particular priority, extension, or context. GotoIf: Conditional goto. GotoIfTime: Conditional Goto based on the current time. Hangup: Hang up the calling channel. IAX2Provision: Provision a calling IAXy with a given template. ICES: Encode and stream using 'ices'. ImportVar: Import a variable from a channel into a new variable. Incomplete: Returns AST_PBX_INCOMPLETE value. JabberJoin: Join a chat room JabberLeave: Leave a chat room JabberSend: Sends an XMPP message to a buddy. JabberSendGroup: Send a Jabber Message to a specified chat room JabberStatus: Retrieve the status of a jabber list member Log: Send arbitrary text to a selected log level. Macro: Macro Implementation. MacroExclusive: Exclusive Macro Implementation. MacroExit: Exit from Macro. MacroIf: Conditional Macro implementation. MailboxExists: Check to see if Voicemail mailbox exists. MeetMe: MeetMe conference bridge. MeetMeAdmin: MeetMe conference administration. MeetMeChannelAdmin: MeetMe conference Administration (channel specific). MeetMeCount: MeetMe participant count. Milliwatt: Generate a Constant 1004Hz tone at 0dbm (mu-law). MinivmAccMess: Record account specific messages. MinivmDelete: Delete Mini-Voicemail voicemail messages. MinivmGreet: Play Mini-Voicemail prompts. MinivmMWI: Send Message Waiting Notification to subscriber(s) of mailbox. MinivmNotify: Notify voicemail owner about new messages. MinivmRecord: Receive Mini-Voicemail and forward via e-mail. MixMonitor: Record a call and mix the audio during the recording. Use of StopMixMonitor is required to guarantee the audio file is available for processing during dialplan execution. Monitor: Monitor a channel. Morsecode: Plays morse code. MP3Player: Play an MP3 file or M3U playlist file or stream. MSet: Set channel variable(s) or function value(s). MusicOnHold: Play Music On Hold indefinitely. MYSQL: Do several mySQLy things NBScat: Play an NBS local stream. NoCDR: Tell Asterisk to not maintain a CDR for the current call NoOp: Do Nothing (No Operation). Originate: Originate a call. Page: Page series of phones Park: Park yourself. ParkAndAnnounce: Park and Announce. ParkedCall: Answer a parked call. PauseMonitor: Pause monitoring of a channel. PauseQueueMember: Pauses a queue member. Pickup: Directed extension call pickup. PickupChan: Pickup a ringing channel. Playback: Play a file. PlayTones: Play a tone list. PrivacyManager: Require phone number to be entered, if no CallerID sent Proceeding: Indicate proceeding. Progress: Indicate progress. Queue: Queue a call for a call queue. QueueLog: Writes to the queue_log file. RaiseException: Handle an exceptional condition. Read: Read a variable. ReadExten: Read an extension into a variable. ReadFile: Read the contents of a text file into a channel variable. ReceiveFAX: Receive a Fax Record: Record to a file. RemoveQueueMember: Dynamically removes queue members. ResetCDR: Resets the Call Data Record. RetryDial: Place a call, retrying on failure allowing an optional exit extension. Return: Return from gosub routine. Ringing: Indicate ringing tone. SayAlpha: Say Alpha. SayDigits: Say Digits. SayNumber: Say Number. SayPhonetic: Say Phonetic. SayUnixTime: Says a specified time in a custom format. SendDTMF: Sends arbitrary DTMF digits SendFAX: Send a Fax SendImage: Sends an image file. SendText: Send a Text Message. SendURL: Send a URL. Set: Set channel variable or function value. SetAMAFlags: Set the AMA Flags. SetCallerPres: Set CallerID Presentation. SetMusicOnHold: Set default Music On Hold class. SIPAddHeader: Add a SIP header to the outbound call. SIPDtmfMode: Change the dtmfmode for a SIP call. SIPRemoveHeader: Remove SIP headers previously added with SIPAddHeader SLAStation: Shared Line Appearance Station. SLATrunk: Shared Line Appearance Trunk. SMS: Communicates with SMS service centres and SMS capable analogue phones. SoftHangup: Hangs up the requested channel. SpeechActivateGrammar: Activate a grammar. SpeechBackground: Play a sound file and wait for speech to be recognized. SpeechCreate: Create a Speech Structure. SpeechDeactivateGrammar: Deactivate a grammar. SpeechDestroy: End speech recognition. SpeechLoadGrammar: Load a grammar. SpeechProcessingSound: Change background processing sound. SpeechStart: Start recognizing voice in the audio stream. SpeechUnloadGrammar: Unload a grammar. StackPop: Remove one address from gosub stack. StartMusicOnHold: Play Music On Hold. StopMixMonitor: Stop recording a call through MixMonitor, and free the recording's file handle. StopMonitor: Stop monitoring a channel. StopMusicOnHold: Stop playing Music On Hold. StopPlayTones: Stop playing a tone list. System: Execute a system command. TestClient: Execute Interface Test Client. TestServer: Execute Interface Test Server. Transfer: Transfer caller to remote extension. TryExec: Executes dialplan application, always returning. TrySystem: Try executing a system command. UnpauseMonitor: Unpause monitoring of a channel. UnpauseQueueMember: Unpauses a queue member. UserEvent: Send an arbitrary event to the manager interface. Verbose: Send arbitrary text to verbose output. VMAuthenticate: Authenticate with Voicemail passwords. VMSayName: Play the name of a voicemail user VoiceMail: Leave a Voicemail message. VoiceMailMain: Check Voicemail messages. Wait: Waits for some time. WaitExten: Waits for an extension to be entered. WaitForNoise: Waits for a specified amount of noise. WaitForRing: Wait for Ring Application. WaitForSilence: Waits for a specified amount of silence. WaitMusicOnHold: Wait, playing Music On Hold. WaitUntil: Wait (sleep) until the current time is the given epoch. While: Start a while loop. Zapateller: Block telemarketers with SIT. -= 181 Applications Registered =- *CLI> |
| safronov_alex wrote: |
| Но вот что мне пишет Астериск [Oct 31 02:37:22] WARNING[1505]: pbx.c:4047 pbx_extension_helper: No application 'Playback ' for extension (6637202_in, 00044404, 4) |
cat /etc/asterisk/modules.conf
_________________
Успехов!
Прописывал в modules.conf
и как load и preload
load => app_playback.so
но все равно
[Oct 31 04:25:05] WARNING[4296]: pbx.c:4047 pbx_extension_helper: No application 'Playback ' for extension (6637202_in, 00044404, 3)
вот мой modules.com
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
preload => app_playback.so
;preload => res_config_odbc.so
;
; Uncomment the following if you wish to use the Speech Recognition API
;preload => res_speech.so
;
; If you want Asterisk to fail if a module does not load, then use
; the "require" keyword. Asterisk will exit with a status code of 2
; if a required module does not load.
;
; require = chan_sip.so
; If you want you can combine with preload
; preload-require = res_odbc.so
;
; If you want, load the GTK console right away.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
;
load => res_musiconhold.so
;load => app_playback.so
;
; Load one of: chan_oss, alsa, or console (portaudio).
; By default, load chan_oss only (automatically).
;
noload => chan_alsa.so
;noload => chan_oss.so
noload => chan_console.so
;
----
но при таком конфиге должны грузиться ВСЕ модули включая playback.
Что не так?
module show like app_playback - проверка загружается ли модуль
попробуйте module load app_playback.so и посмотрите как поругается
ничего не понимаю. Модуль вроде как загружен
server*CLI> module show like app_playback
Module Description Use Count
app_playback.so Sound File Playback Application 0
1 modules loaded
server*CLI> module load app_playback.so
Unable to load module app_playback.so
Command 'module load app_playback.so' failed.
[Oct 31 13:47:53] WARNING[9779]: loader.c:829 load_resource: Module 'app_playback.so' already exists.
Есть идеи?
И еще один вопрос: Можно ли в астериске как нибудь отрегулировать уровень воспроизведения звуковых файлов? (moh, system message etc)
А то орет невыносимо
http://asteriskforum.ru/viewtopic.php?t=7155