QueueMetrics forum
QueueMetrics => General Asterisk configuration => Topic started by: lady_cr0ft on November 08, 2009, 23:16:33
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Hi, we are trying to install TrixBox 2.8 (with Asterisk 1.6), unfortunately
"AddQueueMember" is not executing. We already changed
exten => 25,3,AddQueueMember(${QUEUENAME}|
Local/${AGENTCODE}@from-internal)
exten => 27,3,RemoveQueueMember(${QUEUENAME}|
Local/${AGENTCODE}@from-internal)
TO
exten
=>25,3,AddQueueMember(${QUEUENAME},Local/${AGENTCODE}@from-internal)
exten =>
27,3,RemoveQueueMember(${QUEUENAME},Local/${AGENTCODE}@from-internal)
We have also enabled rewriteLocalChannels:
# Rewrites Local/XXX@ext channels in the format Agent/XXX
default.rewriteLocalChannels=true
We have included this line in extensions_custom.conf:
#include extensions_queuemetrics.conf
But still to no avail. When we checked our CLI we are not getting any
error. We are only getting these:
Manager 'admin' logged on from 127.0.0.1
Manager 'admin' logged off from 127.0.0.1
Here's our AMI credentials in configurations.properties:
cluster.trix.manager=tcp:admin:amp111@127.0.0.1
And our manager.conf:
[admin]
secret = amp111
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
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When you log in to Asterisk, what do you see as "core show applications"?
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-= 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: Attempts 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
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
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
DAHDISendKeypadFacility: Send digits out of band over a PRI
DateTime: Says a specified time in a custom format
DBdel: Delete a key from the database
DBdeltree: Delete a family or keytree from the database
DeadAGI: Executes AGI on a hungup channel
Dial: Place a call and connect to the current channel
Dictate: Virtual Dictation Machine
Directory: Provide directory of voicemail extensions
DISA: DISA (Direct Inward System Access)
DumpChan: Dump Info About The Calling Channel
EAGI: Executes an EAGI compliant application
Echo: Echo audio, video, or 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
JabberSend: JabberSend(jabber,screenname,message)
JabberStatus: JabberStatus(Jabber,ScreenName,Variable)
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
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
Monitor: Monitor a channel
Morsecode: Plays morse code
MP3Player: Play an MP3 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)
NVBackgroundDetect: Background a file with talk and fax detect (IAX and SIP too)
NVFaxDetect: Detects fax sounds on all channel types (IAX and SIP too)
ODBCFinish: Clear the resultset of a successful multirow query
Page: Pages 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 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
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 optional exit extension.
Return: Return from gosub routine
Ringing: Indicate ringing tone
SayAlpha: Say Alpha
SayCountPL: Say Polish counting words
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: Send 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: Soft Hangup Application
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
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
VoiceMail: Leave a Voicemail message
VoiceMailMain: Check Voicemail messages
Wait: Waits for some time
WaitExten: Waits for an extension to be entered
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
-= 172 Applications Registered =-
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As you can see, you have AddQueueMember on your Asterisk system.
This is what I get for AddQueueMember on a 1.6.x:
Connected to Asterisk 1.6.1.6 currently running on pbx-ch (pid = 23691)
-= Info about application 'AddQueueMember' =-
[Synopsis]
Dynamically adds queue members
[Description]
AddQueueMember(queuename[,interface[,penalty[,options[,membername[,stateinterface]]]]]):
Dynamically adds interface to an existing queue.
If the interface is already in the queue it will return an error.
This application sets the following channel variable upon completion:
AQMSTATUS The status of the attempt to add a queue member as a
text string, one of
ADDED | MEMBERALREADY | NOSUCHQUEUE
Example: AddQueueMember(techsupport,SIP/3000)
So it looks like the application and its parameter are correct.
Try and set maximum verbosity and send again a request, to find which error you get.
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:) Here's what I got:
-= Info about application 'AddQueueMember' =-
[Synopsis]
Dynamically adds queue members
[Description]
AddQueueMember(queuename[,interface[,penalty[,options[,membername[,stateinterface]]]]]):
Dynamically adds interface to an existing queue.
If the interface is already in the queue it will return an error.
This application sets the following channel variable upon completion:
AQMSTATUS The status of the attempt to add a queue member as a
text string, one of
ADDED | MEMBERALREADY | NOSUCHQUEUE
Example: AddQueueMember(techsupport,SIP/3000)
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Same as mine ;D
Try and run the command, if it breaks it should log WHY it breaks - see also the /var/log/asterisk/full file for hints...
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??? What do you mean by run the command? Sorry, I'm not yet well versed on these stuff..
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You try to do an AddMember from QM with Asterisk set to maximum verbosity - if it does not work it will print out WHY it does not....
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;D thanks! Here's what's in my /var/log/asterisk/full
[Nov 13 04:02:01] VERBOSE[3563] logger.c: == Parsing '/etc/asterisk/logger.conf': [Nov 13 04:02:01] VERBOSE[3563] logger.c: == Found
[Nov 13 04:02:01] VERBOSE[3563] logger.c: Asterisk Event Logger restarted
[Nov 13 04:02:01] VERBOSE[3563] logger.c: Asterisk Queue Logger restarted
[Nov 13 04:02:02] VERBOSE[3563] logger.c: -- Remote UNIX connection disconnected
[Nov 13 04:56:34] VERBOSE[2520] logger.c: -- Unregistered SIP '5366'
[Nov 13 05:11:03] VERBOSE[2496] logger.c: -- Remote UNIX connection
[Nov 13 05:11:03] VERBOSE[3684] logger.c: -- Remote UNIX connection disconnected
[Nov 13 05:11:03] VERBOSE[2496] logger.c: -- Remote UNIX connection
[Nov 13 05:11:04] VERBOSE[3686] logger.c: -- Remote UNIX connection disconnected
[Nov 13 05:11:04] VERBOSE[2496] logger.c: -- Remote UNIX connection
[Nov 13 05:11:04] VERBOSE[3688] logger.c: -- Remote UNIX connection disconnected
[Nov 13 05:11:04] VERBOSE[2496] logger.c: -- Remote UNIX connection
[Nov 13 05:11:05] VERBOSE[3690] logger.c: -- Remote UNIX connection disconnected
[Nov 13 07:00:47] VERBOSE[2999] logger.c: -- Remote UNIX connection disconnected
[Nov 13 07:22:36] VERBOSE[3724] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Nov 13 07:22:37] VERBOSE[3724] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Nov 13 07:22:37] VERBOSE[3733] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Nov 13 07:22:37] VERBOSE[3733] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Nov 13 07:23:49] VERBOSE[2496] logger.c: -- Remote UNIX connection
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/extconfig.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/manager.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/logger.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: Asterisk Event Logger restarted
[Nov 13 07:23:49] VERBOSE[3736] logger.c: Asterisk Queue Logger restarted
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/features.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/features_general_additional.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/features_general_custom.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/features_applicationmap_additional.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/features_applicationmap_custom.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/features_featuremap_additional.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/features_featuremap_custom.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: -- Added extension '700' priority 1 to parkedcalls (0x9de68c8)
[Nov 13 07:23:49] VERBOSE[3736] logger.c: -- Reloading module 'res_phoneprov.so' (HTTP Phone Provisioning)
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_general_additional.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_general_custom.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_nat.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_registrations_custom.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_registrations.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_custom.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_additional.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/sip_custom_post.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
[Nov 13 07:23:49] VERBOSE[3736] logger.c: == Parsing '/etc/asterisk/phoneprov.conf': [Nov 13 07:23:49] VERBOSE[3736] logger.c: == Found
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I see no trace of a call being sent to it.....
>:(
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Hi
I have exaclty the same problem (trixbox 2.8 with asterisk 1.6.0.10, QM 1.5.5)
the add member button don't seems to work, the only entry I see is following:
Manager 'admin' logged on from 127.0.0.1
Manager 'admin' logged off from 127.0.0.1
Thanks
bye
Moreno
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Did you include the QM dialplan?
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yes
---------snip----------
; This file contains example extensions_custom.conf entries.
; extensions_custom.conf should be used to include customizations
; to AMP's Asterisk dialplan.
; All custom context should contain the string 'custom' in it's name
; Extensions in AMP have access to the 'from-internal' context.
; The context 'from-internal-custom' is included in 'from-internal' by default
#include extensions_queuemetrics.conf
#include extensions_hud.conf
[from-internal-custom]
----------snap-----------
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if you do a dialplan show you see it?
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Yes, i can see the queuemetricis part!
Thanks
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What do you get on the CLI with verbosity 99 when you have QM send a call?
Also: did you define the AMI connection?
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What do you get on the CLI with verbosity 99 when you have QM send a call?
what u mean exctly .. do you mean when i push the add member button
Also: did you define the AMI connection?
where ?
thanks
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1. you have to capture the Asterisk CLI output when you send a commend from QM.
2. See e.g. http://queuemetrics.com/faq.jsp#faq-042-different_manager_port
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1.
Manager 'admin' logged on from 127.0.0.1
Manager 'admin' logged off from 127.0.0.1
2.
callfile.dir=tcp:admin:amp111@127.0.0.1 (trixbox default user/pass)
bye
thanks
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Manager 'admin' logged on from 127.0.0.1
Manager 'admin' logged off from 127.0.0.1
Nothing else? Do you see any error on the QM logs?
Also, try and give your user the ORIGINATE privilege as well.
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FIXED!
with "originate" now works fine!
Grazie
ciao
Moreno
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;D