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Explain the list of commands you can use in extensions.conf or dialplan? jackson 28-July-2008 03:27:30 PMComments AbsoluteTimeout: Set absolute maximum time of call. Deprecated in favor of TIMEOUT(absolute) AddQueueMember: Dynamically adds queue members ADSIProg: Load Asterisk ADSI Scripts into phone AgentCallbackLogin: Call agent callback login. Deprecated. AgentLogin: Call agent login AgentMonitorOutgoing: Monitor Outgoing Agent Calls (0.7.3) AGI: Executes an AGI compliant application AlarmReceiver: Emulate an Ademco Contact ID Alarm Receiver ALSAMonitor: Monitor the ALSA console Answer: Answer a channel if ringing AppendCDRUserField: Append data to the CDR user field. Deprecated in favor of CDR(userfield) Authenticate: Authenticate a user BackGround: Play a file while awaiting extension BackgroundDetect: Background a file with talk detect Bridge: Connect two arbitrary callers (new in Asterisk v1.6) Busy: Indicate busy condition and wait for hangup CallingPres: Change the presentation for the callerid in a ZAP channel ChangeMonitor: Change monitoring filename of a channel ChanIsAvail: Check if channel is available ChannelRedirect: Redirect an existing channel to the dialplan ChanSpy: Universal channel barge-in CheckGroup: checks if the total # of channels exceeds max Congestion: Indicate congestion and wait for hangup ControlPlayback: Play a sound file with fast forward, rewind and exit controls DBdel: Delete a key from the database. DBdeltree: Delete a family or keytree from the database. DBQuery: Execute predefined queries against MySQL Servers, and get the result back into the dialplan. DBRewrite: Execute perl compatible regular expression and substitution out of a MySQL Database. DeadAGI: Executes AGI on a hungup channel Dial: Place an call and connect to the current channel Dictate: Records and plays back a dictation DigitTimeout: Set maximum timeout between digits. Deprecated in favor of TIMEOUT(digit) Directory: Provide directory of voicemail extensions DISA: DISA (Direct Inward System Access) DTMFToText: Enter alphanumeric strings with DTMF phone DUNDiLookup: Look up a number with DUNDi EAGI: Executes an AGI compliant application on local or remote machine (FastAGI) Echo: Echo audio read back to the user EnumLookup: Lookup number in ENUM ExtenSpy: Listen/whisper to a specific extension (new in 1.4) Festival: Say text with the Festival voice synthesizer Flash: Flashes a Zap Trunk Flite: Say text with the Festival Lite voice synthesizer (faster response than Festival) ForkCDR: Fork The CDR into 2 seperate entities GetCPEID: Get ADSI CPE ID GetGroupCount: group count for specified group or channel is in GetGroupMatchCount: Calculates group count for all groups that match pattern Gosub: Jump to a subroutine and return GosubIf: Conditional jump to a subroutine and return Goto: Goto a particular priority, extension, or context GotoIf: Conditional goto GotoIfTime: Conditional goto on current time Hangup: Unconditional hangup HasNewVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. HasVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. ICES: Streaming calls to the Internet ImportVar: Set variable to value JabberStatus: Return presence status of client or transport as values 1-7 JabberSend: Send a message to a buddy LookupBlacklist: Look up Caller*ID name/number from blacklist database. Deprecated in favor of DB. LookupCIDName: Look up CallerID Name from local database. Deprecated in favor of DB. Macro: Macro Implementation MacroExclusive: Only one channel at a time may call this macro, all others have to wait (1.4) MacroExit: Exit the macro as if it had fully completed (1.4) MailboxExists: Checks if voicemail mailbox exists MeetMe: Simple MeetMe conference bridge MeetMeAdmin: MeetMe conference Administration MeetMeCount: MeetMe participant count Milliwatt: Generate a constant 1000Hz tone at 0dbm (mu-law) MiniVM: Mini-Voicemail (new in v1.6) MixMonitor: Record and mix call legs Monitor: Record a telephone conversation to a sound file MP3Player: Play an MP3 sound file or stream MusicOnHold: Play Music On Hold indefinitely MYSQL: Perform various mySQL database activities NBScat: Play an NBS local stream NoCDR: Make sure asterisk doesn't save CDR for a certain call NoOp: No operation. For debugging, see Verbose or Log. Page: Page multiple endpoints at once ParkAndAnnounce: Park and Announce ParkedCall: Answer a parked call PauseQueueMemeber: Pauses an agent Perl: res_perl is the mod_perl of Apache, only for Asterisk Asterisk cmd Pickup: Directed call pickup (in Asterisk 1.2.x) PickUP: Pickup a Zap Channel before answered Playback: Play a file Playtones: Play a tone list while executing other commands PPPD: PPP daemon connector PrivacyManager: Require phone number to be entered, if no CallerID sent Progress: Play early audio to the caller before answering the line Queue: Queue a call for a call queue Read: Read a variable RealTime: Populate variables with details from database using RealTime Record: Record a telephone conversation to a sound file RemoveQueueMember: Dynamically removes queue members ResetCDR: Reset CDR data ResponseTimeout: Set maximum timeout awaiting response. Deprecated in favor of TIMEOUT(response) RetryDial: Place a call, retrying on failure allowing optional exit extension. Return: Return from a Gosub or GosubIf (new in v1.2) Ringing: Indicate ringing tone Rpt: Support Amatuer Radio and Commercial Two Way Repeater Linking SayAlpha: Say Alpha SayDigits: Say Digits SayNumber: Say Number SayPhonetic: Say Phonetic SayUnixTime: Say the date and/or time SendDTMF: Sends arbitrary DTMF digits SendImage: Send an image file SendText: Send client a text message SendURL: Send client a URL to display Set: Set channel variable(s) or function value(s) SetAccount: Sets account code SetAMAflags: Set the channel AMA Flags for billing SetCallerID: Set CallerID. Deprecated in favor of CALLERID. SetCallerPres: Channel independent setting of caller presenation SetCDRUserField: Set CDR User Field. See Billing. Deprecated in favor of CDR(userfield) SetGlobalVar: Set variable to value. Deprecated in favor of GLOBAL. SIPAddHeader: Add header to outbound SIP invite SIPCallPickup: Pickup a ringing phone in the pickup group. SIPGetHeader: Pick any header from a SIP invite message. Deprecated in favor of SIP_HEADER. SIPdtmfMode: Change DTMF mode during SIP call SMS: Send and receive SMS (short messaging service) SoftHangup: Soft Hangup Application SrxEchoCan: Disable/enable Echo Cancellation SrxDeflect: Deflect an incoming call SrxMWI: Set / reset MessageWaitingIndication (MWI) on a Sirrix group Steal: Steal a Zap Channel after answered StackPop: Remove a return address without returning (new in v1.2) StopMonitor: Stop monitoring a channel StopPlaytones: Stop playing a tone list System: Execute a system command Transfer: Transfer caller to remote extension TestClient: Execute Interface Test Client TestServer: Execute Interface Test Server TrySystem: Execute a system command with always 0 returned TXTCIDName: Lookup caller name from TXT record UnpauseQueueMemeber: Resumes an agent UserEvent: Send an arbitrary event to the manager interface VMAuthenticate: Authenticate a user based on voicemail.conf VoiceMail: Leave a voicemail message VoiceMailMain: Enter voicemail system Wait: Waits for some time WaitExten: Waits for some time WaitForRing: Wait for Ring Application WaitMusicOnHold: Wait, playing Music On Hold Zapateller: Block telemarketers with SIT ZapBarge: Barge in (monitor) Zap channel ZapSendKeypadFacility: Send digits out of band over a PRI ZapRAS: Provide ISDN data service ZapScan: Scan Zap channels to monitor calls Posted by crouse The extensions.conf file is one of the most used and most important configuration file in Asterisk PBX - it contains the dialplan. What is a dialplan? The dialplan , or we can say "the heart of the Asterisk System", defines how Asterisk PBX will handle incoming and outgoing calls, it also contains all extension numbers. The dialplan is divided in sections called contexts. Every context consists from more than one extensions. What is an extension? The extension is the telephone number, it can be numbers, letters or both. Every extension has a priority and an application. With the help of contexts we can organize our dialplan. he "extensions.conf" file is made up of sections or contexts between brackets [ ] There are always two special contexts; [ general ] and [ globals] [general] context The [general] manages a few general options: - static: It indicates if a "save dialplan" command from the console is possible. By default "yes". It works altogether with "writeprotect" - writeprotect: If writeprotect=no and static=yes "save dialplan" command from the console is possible. The default value is "no". - autofallthrough: If it is activated and an extension is without things to do the call is finished with a BUSY, CONGESTION or HANGUP message If it is not activated it will be waiting for another extension. It is not convenient that an extension remains without things to do as we explain later. - clearglobalvars: If it is activated Asterisk release the global variables when the extensions are recharged or when Asterisk is restarted. - priorityjumping: with 'yes' value the application supports ' jumping' or jump to different priorities. Deprecated [globals] context In this context global variables are defined and can be used in the rest of contexts. For example CONSOLE=Console/dsp ; when we use CONSOLE word we are calling to /Console/dsp Variables usually are always in capital letters to differentiate them later. Other contexts[] We are going to show now how to create an specific context and to assign it a dialplan. All lines of a certain context have the same format: exten => extension , priority, Command(parameters) Extensión is the caller number Priority is the order that commands are run. First the one with priority 1, then with 2, ... Command is the thing to do. We are going to learn the commands with some examples. Example 1: Hangup exten => 333,1,Hangup ; when someone calls the number 333 the priority 1 is executed and the system makes a hangup Example 2 : Call to 3000 SIP user and if it is not available call the voicemail. exten => 3000,1,Dial(SIP/3000,30,Ttm) ; call to the 3000 SIP user that must be defined in sip.conf with that context exten => 3000,2,Hangup ; hangup when the call finishs exten => 3000,102,Voicemail(3000) ; 102 Priority is when the user is not connected and the voicemail number 3000 starts. exten => 3000,103,Hangup ; hangup when the message is left To call to 3000 extension we use Dial(destination,timeout time,options) command The destination is the user 3000 of the file sip.conf, we have a timeout of 30 seconds. The user 3000 should exist in sip.conf The options are options of the dial command: "T" allows the user to transfer the call pressing # "t" allows the user to transfer the call pressing # "m" puts music on hold while we are waiting the other user to respond. You can try without it If user 3000 is not connected the systems goes to the actual priority + 101 if it exists (in this case 102) and the voicemail number 3000 starts. It is very important that all the branches finish with a hangup command. Example 3 : Echo and latency exten => 600,1,Playback(demo-echotest) ; Sound of echotest exten => 600,2,Echo ; echo test is run exten => 600,3,Playback(demo-echodone) ; sound what we said exten => 600,4,Hangup ; hangup We call number 600 and the things we said are going to be repeated. We can test the latency of the system Example 4 : Extension "start" exten => s,1,Wait,1 ; Wait a second exten => s,2,Answer ; Answer. Asterisk itself get the call exten => s,3,DigitTimeout,5 ; Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,10 ; Response Timeout to 10 seconds exten => s,5,BackGround(demo-congrats) ; A sound archive exten => s,6,hangup ; hangup exten => 1000,1,Goto(mycontext,s,1) ; When we call 1000 number it goes to s extension with priority 1 of "mycontext" In this case we learn the s (start) extension. It is the one that takes the calls when we are in this context but do not know the extension. Also, it is possible to enter from another extension, for example in this case calling to 1000 extension. With Goto we can go to the context, extension and priority that we want. Example 5 : Call to a Voip SIP provider exten => _340.,1,Dial(SIP/${EXTEN:3}@sipprovider,90,Tt) exten => _340.,2,hangup ; hangup exten => _20.,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) exten => _20.,2,hangup ; hangup In this case what we do is that when we call 340 followed by any number (340 is the prefix) we will call to a SIP number. For example in the first case if we press 340600600 number we are calling to the 600600 number of the sipprovider defined in sip.conf. (EXTEN:3 means that we deleted the three first numbers) In the second case if we also press 2060600 we will be calling to the same one number "600600" of "sipprovider" (EXTEN:2) In the previous cases the point . replaces any character but we can also use X - number from 0 to 9 Z - number from 1 to 9 N - number from 2 to 9 [1,5-7] - number 1, 5, 6 or 7 exten => _20XX,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) ; we must call 20 and two numbers more (no characters) exten => _20ZZ.,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) ; we must call 20, and two numbers from 1 to 9 and anything more exten => _20[1-3]..,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) ; we must call 20, a number from 1 to 3 and anything more Posted by sagitraz AbsoluteTimeout: Set absolute maximum time of call. Deprecated in favor of TIMEOUT(absolute) AddQueueMember: Dynamically adds queue members ADSIProg: Load Asterisk ADSI Scripts into phone AgentCallbackLogin: Call agent callback login. Deprecated. AgentLogin: Call agent login AgentMonitorOutgoing: Monitor Outgoing Agent Calls (0.7.3) AGI: Executes an AGI compliant application AlarmReceiver: Emulate an Ademco Contact ID Alarm Receiver ALSAMonitor: Monitor the ALSA console Answer: Answer a channel if ringing AppendCDRUserField: Append data to the CDR user field. Deprecated in favor of CDR(userfield) Authenticate: Authenticate a user BackGround: Play a file while awaiting extension BackgroundDetect: Background a file with talk detect Bridge: Connect two arbitrary callers (new in Asterisk v1.6) Busy: Indicate busy condition and wait for hangup CallingPres: Change the presentation for the callerid in a ZAP channel ChangeMonitor: Change monitoring filename of a channel ChanIsAvail: Check if channel is available ChannelRedirect: Redirect an existing channel to the dialplan ChanSpy: Universal channel barge-in CheckGroup: checks if the total # of channels exceeds max Congestion: Indicate congestion and wait for hangup ControlPlayback: Play a sound file with fast forward, rewind and exit controls DBdel: Delete a key from the database. DBdeltree: Delete a family or keytree from the database. DBQuery: Execute predefined queries against MySQL Servers, and get the result back into the dialplan. DBRewrite: Execute perl compatible regular expression and substitution out of a MySQL Database. DeadAGI: Executes AGI on a hungup channel Dial: Place an call and connect to the current channel Dictate: Records and plays back a dictation DigitTimeout: Set maximum timeout between digits. Deprecated in favor of TIMEOUT(digit) Directory: Provide directory of voicemail extensions DISA: DISA (Direct Inward System Access) DTMFToText: Enter alphanumeric strings with DTMF phone DUNDiLookup: Look up a number with DUNDi EAGI: Executes an AGI compliant application on local or remote machine (FastAGI) Echo: Echo audio read back to the user EnumLookup: Lookup number in ENUM ExtenSpy: Listen/whisper to a specific extension (new in 1.4) Festival: Say text with the Festival voice synthesizer Flash: Flashes a Zap Trunk Flite: Say text with the Festival Lite voice synthesizer (faster response than Festival) ForkCDR: Fork The CDR into 2 seperate entities GetCPEID: Get ADSI CPE ID GetGroupCount: group count for specified group or channel is in GetGroupMatchCount: Calculates group count for all groups that match pattern Gosub: Jump to a subroutine and return GosubIf: Conditional jump to a subroutine and return Goto: Goto a particular priority, extension, or context GotoIf: Conditional goto GotoIfTime: Conditional goto on current time Hangup: Unconditional hangup HasNewVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. HasVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. ICES: Streaming calls to the Internet ImportVar: Set variable to value JabberStatus: Return presence status of client or transport as values 1-7 JabberSend: Send a message to a buddy LookupBlacklist: Look up Caller*ID name/number from blacklist database. Deprecated in favor of DB. LookupCIDName: Look up CallerID Name from local database. Deprecated in favor of DB. Macro: Macro Implementation MacroExclusive: Only one channel at a time may call this macro, all others have to wait (1.4) MacroExit: Exit the macro as if it had fully completed (1.4) MailboxExists: Checks if voicemail mailbox exists MeetMe: Simple MeetMe conference bridge MeetMeAdmin: MeetMe conference Administration MeetMeCount: MeetMe participant count Milliwatt: Generate a constant 1000Hz tone at 0dbm (mu-law) MiniVM: Mini-Voicemail (new in v1.6) MixMonitor: Record and mix call legs Monitor: Record a telephone conversation to a sound file MP3Player: Play an MP3 sound file or stream MusicOnHold: Play Music On Hold indefinitely MYSQL: Perform various mySQL database activities NBScat: Play an NBS local stream NoCDR: Make sure asterisk doesn't save CDR for a certain call NoOp: No operation. For debugging, see Verbose or Log. Page: Page multiple endpoints at once ParkAndAnnounce: Park and Announce ParkedCall: Answer a parked call PauseQueueMemeber: Pauses an agent Perl: res_perl is the mod_perl of Apache, only for Asterisk Asterisk cmd Pickup: Directed call pickup (in Asterisk 1.2.x) PickUP: Pickup a Zap Channel before answered Playback: Play a file Playtones: Play a tone list while executing other commands PPPD: PPP daemon connector PrivacyManager: Require phone number to be entered, if no CallerID sent Progress: Play early audio to the caller before answering the line Queue: Queue a call for a call queue Read: Read a variable RealTime: Populate variables with details from database using RealTime Record: Record a telephone conversation to a sound file RemoveQueueMember: Dynamically removes queue members ResetCDR: Reset CDR data ResponseTimeout: Set maximum timeout awaiting response. Deprecated in favor of TIMEOUT(response) RetryDial: Place a call, retrying on failure allowing optional exit extension. Return: Return from a Gosub or GosubIf (new in v1.2) Ringing: Indicate ringing tone Rpt: Support Amatuer Radio and Commercial Two Way Repeater Linking SayAlpha: Say Alpha SayDigits: Say Digits SayNumber: Say Number SayPhonetic: Say Phonetic SayUnixTime: Say the date and/or time SendDTMF: Sends arbitrary DTMF digits SendImage: Send an image file SendText: Send client a text message SendURL: Send client a URL to display Set: Set channel variable(s) or function value(s) SetAccount: Sets account code SetAMAflags: Set the channel AMA Flags for billing SetCallerID: Set CallerID. Deprecated in favor of CALLERID. SetCallerPres: Channel independent setting of caller presenation SetCDRUserField: Set CDR User Field. See Billing. Deprecated in favor of CDR(userfield) SetGlobalVar: Set variable to value. Deprecated in favor of GLOBAL. SIPAddHeader: Add header to outbound SIP invite SIPCallPickup: Pickup a ringing phone in the pickup group. SIPGetHeader: Pick any header from a SIP invite message. Deprecated in favor of SIP_HEADER. SIPdtmfMode: Change DTMF mode during SIP call SMS: Send and receive SMS (short messaging service) SoftHangup: Soft Hangup Application SrxEchoCan: Disable/enable Echo Cancellation SrxDeflect: Deflect an incoming call SrxMWI: Set / reset MessageWaitingIndication (MWI) on a Sirrix group Steal: Steal a Zap Channel after answered StackPop: Remove a return address without returning (new in v1.2) StopMonitor: Stop monitoring a channel StopPlaytones: Stop playing a tone list System: Execute a system command Transfer: Transfer caller to remote extension TestClient: Execute Interface Test Client TestServer: Execute Interface Test Server TrySystem: Execute a system command with always 0 returned TXTCIDName: Lookup caller name from TXT record UnpauseQueueMemeber: Resumes an agent UserEvent: Send an arbitrary event to the manager interface VMAuthenticate: Authenticate a user based on voicemail.conf VoiceMail: Leave a voicemail message VoiceMailMain: Enter voicemail system Wait: Waits for some time WaitExten: Waits for some time WaitForRing: Wait for Ring Application WaitMusicOnHold: Wait, playing Music On Hold Zapateller: Block telemarketers with SIT ZapBarge: Barge in (monitor) Zap channel ZapSendKeypadFacility: Send digits out of band over a PRI ZapRAS: Provide ISDN data service ZapScan: Scan Zap channels to monitor calls Posted by william |
Posted: 29-July-2008 01:12:20 PM By: william AbsoluteTimeout: Set absolute maximum time of call. Deprecated in favor of TIMEOUT(absolute) AddQueueMember: Dynamically adds queue members ADSIProg: Load Asterisk ADSI Scripts into phone AgentCallbackLogin: Call agent callback login. Deprecated. AgentLogin: Call agent login AgentMonitorOutgoing: Monitor Outgoing Agent Calls (0.7.3) AGI: Executes an AGI compliant application AlarmReceiver: Emulate an Ademco Contact ID Alarm Receiver ALSAMonitor: Monitor the ALSA console Answer: Answer a channel if ringing AppendCDRUserField: Append data to the CDR user field. Deprecated in favor of CDR(userfield) Authenticate: Authenticate a user BackGround: Play a file while awaiting extension BackgroundDetect: Background a file with talk detect Bridge: Connect two arbitrary callers (new in Asterisk v1.6) Busy: Indicate busy condition and wait for hangup CallingPres: Change the presentation for the callerid in a ZAP channel ChangeMonitor: Change monitoring filename of a channel ChanIsAvail: Check if channel is available ChannelRedirect: Redirect an existing channel to the dialplan ChanSpy: Universal channel barge-in CheckGroup: checks if the total # of channels exceeds max Congestion: Indicate congestion and wait for hangup ControlPlayback: Play a sound file with fast forward, rewind and exit controls DBdel: Delete a key from the database. DBdeltree: Delete a family or keytree from the database. DBQuery: Execute predefined queries against MySQL Servers, and get the result back into the dialplan. DBRewrite: Execute perl compatible regular expression and substitution out of a MySQL Database. DeadAGI: Executes AGI on a hungup channel Dial: Place an call and connect to the current channel Dictate: Records and plays back a dictation DigitTimeout: Set maximum timeout between digits. Deprecated in favor of TIMEOUT(digit) Directory: Provide directory of voicemail extensions DISA: DISA (Direct Inward System Access) DTMFToText: Enter alphanumeric strings with DTMF phone DUNDiLookup: Look up a number with DUNDi EAGI: Executes an AGI compliant application on local or remote machine (FastAGI) Echo: Echo audio read back to the user EnumLookup: Lookup number in ENUM ExtenSpy: Listen/whisper to a specific extension (new in 1.4) Festival: Say text with the Festival voice synthesizer Flash: Flashes a Zap Trunk Flite: Say text with the Festival Lite voice synthesizer (faster response than Festival) ForkCDR: Fork The CDR into 2 seperate entities GetCPEID: Get ADSI CPE ID GetGroupCount: group count for specified group or channel is in GetGroupMatchCount: Calculates group count for all groups that match pattern Gosub: Jump to a subroutine and return GosubIf: Conditional jump to a subroutine and return Goto: Goto a particular priority, extension, or context GotoIf: Conditional goto GotoIfTime: Conditional goto on current time Hangup: Unconditional hangup HasNewVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. HasVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. ICES: Streaming calls to the Internet ImportVar: Set variable to value JabberStatus: Return presence status of client or transport as values 1-7 JabberSend: Send a message to a buddy LookupBlacklist: Look up Caller*ID name/number from blacklist database. Deprecated in favor of DB. LookupCIDName: Look up CallerID Name from local database. Deprecated in favor of DB. Macro: Macro Implementation MacroExclusive: Only one channel at a time may call this macro, all others have to wait (1.4) MacroExit: Exit the macro as if it had fully completed (1.4) MailboxExists: Checks if voicemail mailbox exists MeetMe: Simple MeetMe conference bridge MeetMeAdmin: MeetMe conference Administration MeetMeCount: MeetMe participant count Milliwatt: Generate a constant 1000Hz tone at 0dbm (mu-law) MiniVM: Mini-Voicemail (new in v1.6) MixMonitor: Record and mix call legs Monitor: Record a telephone conversation to a sound file MP3Player: Play an MP3 sound file or stream MusicOnHold: Play Music On Hold indefinitely MYSQL: Perform various mySQL database activities NBScat: Play an NBS local stream NoCDR: Make sure asterisk doesn't save CDR for a certain call NoOp: No operation. For debugging, see Verbose or Log. Page: Page multiple endpoints at once ParkAndAnnounce: Park and Announce ParkedCall: Answer a parked call PauseQueueMemeber: Pauses an agent Perl: res_perl is the mod_perl of Apache, only for Asterisk Asterisk cmd Pickup: Directed call pickup (in Asterisk 1.2.x) PickUP: Pickup a Zap Channel before answered Playback: Play a file Playtones: Play a tone list while executing other commands PPPD: PPP daemon connector PrivacyManager: Require phone number to be entered, if no CallerID sent Progress: Play early audio to the caller before answering the line Queue: Queue a call for a call queue Read: Read a variable RealTime: Populate variables with details from database using RealTime Record: Record a telephone conversation to a sound file RemoveQueueMember: Dynamically removes queue members ResetCDR: Reset CDR data ResponseTimeout: Set maximum timeout awaiting response. Deprecated in favor of TIMEOUT(response) RetryDial: Place a call, retrying on failure allowing optional exit extension. Return: Return from a Gosub or GosubIf (new in v1.2) Ringing: Indicate ringing tone Rpt: Support Amatuer Radio and Commercial Two Way Repeater Linking SayAlpha: Say Alpha SayDigits: Say Digits SayNumber: Say Number SayPhonetic: Say Phonetic SayUnixTime: Say the date and/or time SendDTMF: Sends arbitrary DTMF digits SendImage: Send an image file SendText: Send client a text message SendURL: Send client a URL to display Set: Set channel variable(s) or function value(s) SetAccount: Sets account code SetAMAflags: Set the channel AMA Flags for billing SetCallerID: Set CallerID. Deprecated in favor of CALLERID. SetCallerPres: Channel independent setting of caller presenation SetCDRUserField: Set CDR User Field. See Billing. Deprecated in favor of CDR(userfield) SetGlobalVar: Set variable to value. Deprecated in favor of GLOBAL. SIPAddHeader: Add header to outbound SIP invite SIPCallPickup: Pickup a ringing phone in the pickup group. SIPGetHeader: Pick any header from a SIP invite message. Deprecated in favor of SIP_HEADER. SIPdtmfMode: Change DTMF mode during SIP call SMS: Send and receive SMS (short messaging service) SoftHangup: Soft Hangup Application SrxEchoCan: Disable/enable Echo Cancellation SrxDeflect: Deflect an incoming call SrxMWI: Set / reset MessageWaitingIndication (MWI) on a Sirrix group Steal: Steal a Zap Channel after answered StackPop: Remove a return address without returning (new in v1.2) StopMonitor: Stop monitoring a channel StopPlaytones: Stop playing a tone list System: Execute a system command Transfer: Transfer caller to remote extension TestClient: Execute Interface Test Client TestServer: Execute Interface Test Server TrySystem: Execute a system command with always 0 returned TXTCIDName: Lookup caller name from TXT record UnpauseQueueMemeber: Resumes an agent UserEvent: Send an arbitrary event to the manager interface VMAuthenticate: Authenticate a user based on voicemail.conf VoiceMail: Leave a voicemail message VoiceMailMain: Enter voicemail system Wait: Waits for some time WaitExten: Waits for some time WaitForRing: Wait for Ring Application WaitMusicOnHold: Wait, playing Music On Hold Zapateller: Block telemarketers with SIT ZapBarge: Barge in (monitor) Zap channel ZapSendKeypadFacility: Send digits out of band over a PRI ZapRAS: Provide ISDN data service ZapScan: Scan Zap channels to monitor calls | |
Posted: 24-August-2008 08:13:14 AM By: sagitraz The extensions.conf file is one of the most used and most important configuration file in Asterisk PBX - it contains the dialplan. What is a dialplan? The dialplan , or we can say "the heart of the Asterisk System", defines how Asterisk PBX will handle incoming and outgoing calls, it also contains all extension numbers. The dialplan is divided in sections called contexts. Every context consists from more than one extensions. What is an extension? The extension is the telephone number, it can be numbers, letters or both. Every extension has a priority and an application. With the help of contexts we can organize our dialplan. he "extensions.conf" file is made up of sections or contexts between brackets [ ] There are always two special contexts; [ general ] and [ globals] [general] context The [general] manages a few general options: - static: It indicates if a "save dialplan" command from the console is possible. By default "yes". It works altogether with "writeprotect" - writeprotect: If writeprotect=no and static=yes "save dialplan" command from the console is possible. The default value is "no". - autofallthrough: If it is activated and an extension is without things to do the call is finished with a BUSY, CONGESTION or HANGUP message If it is not activated it will be waiting for another extension. It is not convenient that an extension remains without things to do as we explain later. - clearglobalvars: If it is activated Asterisk release the global variables when the extensions are recharged or when Asterisk is restarted. - priorityjumping: with 'yes' value the application supports ' jumping' or jump to different priorities. Deprecated [globals] context In this context global variables are defined and can be used in the rest of contexts. For example CONSOLE=Console/dsp ; when we use CONSOLE word we are calling to /Console/dsp Variables usually are always in capital letters to differentiate them later. Other contexts[] We are going to show now how to create an specific context and to assign it a dialplan. All lines of a certain context have the same format: exten => extension , priority, Command(parameters) Extensión is the caller number Priority is the order that commands are run. First the one with priority 1, then with 2, ... Command is the thing to do. We are going to learn the commands with some examples. Example 1: Hangup exten => 333,1,Hangup ; when someone calls the number 333 the priority 1 is executed and the system makes a hangup Example 2 : Call to 3000 SIP user and if it is not available call the voicemail. exten => 3000,1,Dial(SIP/3000,30,Ttm) ; call to the 3000 SIP user that must be defined in sip.conf with that context exten => 3000,2,Hangup ; hangup when the call finishs exten => 3000,102,Voicemail(3000) ; 102 Priority is when the user is not connected and the voicemail number 3000 starts. exten => 3000,103,Hangup ; hangup when the message is left To call to 3000 extension we use Dial(destination,timeout time,options) command The destination is the user 3000 of the file sip.conf, we have a timeout of 30 seconds. The user 3000 should exist in sip.conf The options are options of the dial command: "T" allows the user to transfer the call pressing # "t" allows the user to transfer the call pressing # "m" puts music on hold while we are waiting the other user to respond. You can try without it If user 3000 is not connected the systems goes to the actual priority + 101 if it exists (in this case 102) and the voicemail number 3000 starts. It is very important that all the branches finish with a hangup command. Example 3 : Echo and latency exten => 600,1,Playback(demo-echotest) ; Sound of echotest exten => 600,2,Echo ; echo test is run exten => 600,3,Playback(demo-echodone) ; sound what we said exten => 600,4,Hangup ; hangup We call number 600 and the things we said are going to be repeated. We can test the latency of the system Example 4 : Extension "start" exten => s,1,Wait,1 ; Wait a second exten => s,2,Answer ; Answer. Asterisk itself get the call exten => s,3,DigitTimeout,5 ; Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,10 ; Response Timeout to 10 seconds exten => s,5,BackGround(demo-congrats) ; A sound archive exten => s,6,hangup ; hangup exten => 1000,1,Goto(mycontext,s,1) ; When we call 1000 number it goes to s extension with priority 1 of "mycontext" In this case we learn the s (start) extension. It is the one that takes the calls when we are in this context but do not know the extension. Also, it is possible to enter from another extension, for example in this case calling to 1000 extension. With Goto we can go to the context, extension and priority that we want. Example 5 : Call to a Voip SIP provider exten => _340.,1,Dial(SIP/${EXTEN:3}@sipprovider,90,Tt) exten => _340.,2,hangup ; hangup exten => _20.,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) exten => _20.,2,hangup ; hangup In this case what we do is that when we call 340 followed by any number (340 is the prefix) we will call to a SIP number. For example in the first case if we press 340600600 number we are calling to the 600600 number of the sipprovider defined in sip.conf. (EXTEN:3 means that we deleted the three first numbers) In the second case if we also press 2060600 we will be calling to the same one number "600600" of "sipprovider" (EXTEN:2) In the previous cases the point . replaces any character but we can also use X - number from 0 to 9 Z - number from 1 to 9 N - number from 2 to 9 [1,5-7] - number 1, 5, 6 or 7 exten => _20XX,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) ; we must call 20 and two numbers more (no characters) exten => _20ZZ.,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) ; we must call 20, and two numbers from 1 to 9 and anything more exten => _20[1-3]..,1,Dial(SIP/${EXTEN:2}@sipprovider,90,Tt) ; we must call 20, a number from 1 to 3 and anything more | |
Posted: 01-March-2009 07:12:55 AM By: crouse AbsoluteTimeout: Set absolute maximum time of call. Deprecated in favor of TIMEOUT(absolute) AddQueueMember: Dynamically adds queue members ADSIProg: Load Asterisk ADSI Scripts into phone AgentCallbackLogin: Call agent callback login. Deprecated. AgentLogin: Call agent login AgentMonitorOutgoing: Monitor Outgoing Agent Calls (0.7.3) AGI: Executes an AGI compliant application AlarmReceiver: Emulate an Ademco Contact ID Alarm Receiver ALSAMonitor: Monitor the ALSA console Answer: Answer a channel if ringing AppendCDRUserField: Append data to the CDR user field. Deprecated in favor of CDR(userfield) Authenticate: Authenticate a user BackGround: Play a file while awaiting extension BackgroundDetect: Background a file with talk detect Bridge: Connect two arbitrary callers (new in Asterisk v1.6) Busy: Indicate busy condition and wait for hangup CallingPres: Change the presentation for the callerid in a ZAP channel ChangeMonitor: Change monitoring filename of a channel ChanIsAvail: Check if channel is available ChannelRedirect: Redirect an existing channel to the dialplan ChanSpy: Universal channel barge-in CheckGroup: checks if the total # of channels exceeds max Congestion: Indicate congestion and wait for hangup ControlPlayback: Play a sound file with fast forward, rewind and exit controls DBdel: Delete a key from the database. DBdeltree: Delete a family or keytree from the database. DBQuery: Execute predefined queries against MySQL Servers, and get the result back into the dialplan. DBRewrite: Execute perl compatible regular expression and substitution out of a MySQL Database. DeadAGI: Executes AGI on a hungup channel Dial: Place an call and connect to the current channel Dictate: Records and plays back a dictation DigitTimeout: Set maximum timeout between digits. Deprecated in favor of TIMEOUT(digit) Directory: Provide directory of voicemail extensions DISA: DISA (Direct Inward System Access) DTMFToText: Enter alphanumeric strings with DTMF phone DUNDiLookup: Look up a number with DUNDi EAGI: Executes an AGI compliant application on local or remote machine (FastAGI) Echo: Echo audio read back to the user EnumLookup: Lookup number in ENUM ExtenSpy: Listen/whisper to a specific extension (new in 1.4) Festival: Say text with the Festival voice synthesizer Flash: Flashes a Zap Trunk Flite: Say text with the Festival Lite voice synthesizer (faster response than Festival) ForkCDR: Fork The CDR into 2 seperate entities GetCPEID: Get ADSI CPE ID GetGroupCount: group count for specified group or channel is in GetGroupMatchCount: Calculates group count for all groups that match pattern Gosub: Jump to a subroutine and return GosubIf: Conditional jump to a subroutine and return Goto: Goto a particular priority, extension, or context GotoIf: Conditional goto GotoIfTime: Conditional goto on current time Hangup: Unconditional hangup HasNewVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. HasVoicemail: Conditionally branches to priority + 101. Deprecated in favor of VMCOUNT. ICES: Streaming calls to the Internet ImportVar: Set variable to value JabberStatus: Return presence status of client or transport as values 1-7 JabberSend: Send a message to a buddy LookupBlacklist: Look up Caller*ID name/number from blacklist database. Deprecated in favor of DB. LookupCIDName: Look up CallerID Name from local database. Deprecated in favor of DB. Macro: Macro Implementation MacroExclusive: Only one channel at a time may call this macro, all others have to wait (1.4) MacroExit: Exit the macro as if it had fully completed (1.4) MailboxExists: Checks if voicemail mailbox exists MeetMe: Simple MeetMe conference bridge MeetMeAdmin: MeetMe conference Administration MeetMeCount: MeetMe participant count Milliwatt: Generate a constant 1000Hz tone at 0dbm (mu-law) MiniVM: Mini-Voicemail (new in v1.6) MixMonitor: Record and mix call legs Monitor: Record a telephone conversation to a sound file MP3Player: Play an MP3 sound file or stream MusicOnHold: Play Music On Hold indefinitely MYSQL: Perform various mySQL database activities NBScat: Play an NBS local stream NoCDR: Make sure asterisk doesn't save CDR for a certain call NoOp: No operation. For debugging, see Verbose or Log. Page: Page multiple endpoints at once ParkAndAnnounce: Park and Announce ParkedCall: Answer a parked call PauseQueueMemeber: Pauses an agent Perl: res_perl is the mod_perl of Apache, only for Asterisk Asterisk cmd Pickup: Directed call pickup (in Asterisk 1.2.x) PickUP: Pickup a Zap Channel before answered Playback: Play a file Playtones: Play a tone list while executing other commands PPPD: PPP daemon connector PrivacyManager: Require phone number to be entered, if no CallerID sent Progress: Play early audio to the caller before answering the line Queue: Queue a call for a call queue Read: Read a variable RealTime: Populate variables with details from database using RealTime Record: Record a telephone conversation to a sound file RemoveQueueMember: Dynamically removes queue members ResetCDR: Reset CDR data ResponseTimeout: Set maximum timeout awaiting response. Deprecated in favor of TIMEOUT(response) RetryDial: Place a call, retrying on failure allowing optional exit extension. Return: Return from a Gosub or GosubIf (new in v1.2) Ringing: Indicate ringing tone Rpt: Support Amatuer Radio and Commercial Two Way Repeater Linking SayAlpha: Say Alpha SayDigits: Say Digits SayNumber: Say Number SayPhonetic: Say Phonetic SayUnixTime: Say the date and/or time SendDTMF: Sends arbitrary DTMF digits SendImage: Send an image file SendText: Send client a text message SendURL: Send client a URL to display Set: Set channel variable(s) or function value(s) SetAccount: Sets account code SetAMAflags: Set the channel AMA Flags for billing SetCallerID: Set CallerID. Deprecated in favor of CALLERID. SetCallerPres: Channel independent setting of caller presenation SetCDRUserField: Set CDR User Field. See Billing. Deprecated in favor of CDR(userfield) SetGlobalVar: Set variable to value. Deprecated in favor of GLOBAL. SIPAddHeader: Add header to outbound SIP invite SIPCallPickup: Pickup a ringing phone in the pickup group. SIPGetHeader: Pick any header from a SIP invite message. Deprecated in favor of SIP_HEADER. SIPdtmfMode: Change DTMF mode during SIP call SMS: Send and receive SMS (short messaging service) SoftHangup: Soft Hangup Application SrxEchoCan: Disable/enable Echo Cancellation SrxDeflect: Deflect an incoming call SrxMWI: Set / reset MessageWaitingIndication (MWI) on a Sirrix group Steal: Steal a Zap Channel after answered StackPop: Remove a return address without returning (new in v1.2) StopMonitor: Stop monitoring a channel StopPlaytones: Stop playing a tone list System: Execute a system command Transfer: Transfer caller to remote extension TestClient: Execute Interface Test Client TestServer: Execute Interface Test Server TrySystem: Execute a system command with always 0 returned TXTCIDName: Lookup caller name from TXT record UnpauseQueueMemeber: Resumes an agent UserEvent: Send an arbitrary event to the manager interface VMAuthenticate: Authenticate a user based on voicemail.conf VoiceMail: Leave a voicemail message VoiceMailMain: Enter voicemail system Wait: Waits for some time WaitExten: Waits for some time WaitForRing: Wait for Ring Application WaitMusicOnHold: Wait, playing Music On Hold Zapateller: Block telemarketers with SIT ZapBarge: Barge in (monitor) Zap channel ZapSendKeypadFacility: Send digits out of band over a PRI ZapRAS: Provide ISDN data service ZapScan: Scan Zap channels to monitor calls |