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Kubernetes dynamic configuration engine for Asterisk

License: Other

Shell 3.14% Go 95.18% Dockerfile 1.69%

asterisk go golang kubernetes

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asterisk-config's Issues

Migrate to mainline k8s client

The ericchiang client we currently use was great for its time, but now that the kubernetes client is split off from the main package, we should use it instead.

Re-render is not triggered on secret value change

When changing the zip file in the secret that is mounted in the asterisk-config volume, it does not detect it and re-renders the template. This makes it hard to have dynamic configuration except from the already implemented reactivity based on the network changes.

Possible solutions:

Watch the file system for changes on the mounted secret and re-renders the full config and overwrite the original (this have a drawback of taking some time, depending on the cluster, to update due to https://kubernetes.io/docs/concepts/configuration/secret/#mounted-secrets-are-updated-automatically)
Watch for secret changes via k8s API and trigger the re-render the same way

Race condition on asterisk custom configuration

I have been able to successfully deploy Kamailio and Asterisk with asterisk-config as a sidecar in GCP picking up the custom configuration files from a kubernetes secret. The overall setup is working fine (thanks for that Sean!) but occasionally asterisk-config container fails and the pod is restarted. Once the pod is restarted, asterisk fails to load the custom modules.conf stored in the K8s secret and instead, it loads the default modules.conf that is shipped with the cycore image hosted at Dockerhub.

These are the startup logs for asterisk-config. It seems to be detecting lots of changes even though nothing has changed and maybe this is causing the issue.

2018/10/08 21:23:34 WARNING: unhandled cloud
2018/10/08 21:23:34 running service
2018/10/08 21:23:34 change detected
2018/10/08 21:23:34 service exited: failed to reload asterisk modules: failed to contact ARI to reload module res_pjsip.so: Put http://127.0.0.1:8088/ari/asterisk/modules/res_pjsip.so: dial tcp 127.0.0.1:8088: connect: connection refused
2018/10/08 21:23:34 running service
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 service exited: failed to reload asterisk modules: failed to contact ARI to reload module res_pjsip.so: Put http://127.0.0.1:8088/ari/asterisk/modules/res_pjsip.so: dial tcp 127.0.0.1:8088: connect: connection refused
2018/10/08 21:23:35 running service
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 service exited: failed to reload asterisk modules: module res_pjsip.so not already loaded
2018/10/08 21:23:35 running service
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 service exited: failed to reload asterisk modules: module res_pjsip.so reload failed: 503 Service Unavailable
2018/10/08 21:23:35 running service
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 reloads complete
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 reloads complete
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 reloads complete
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 reloads complete
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 reloads complete
2018/10/08 21:23:35 change detected
2018/10/08 21:23:35 reloads complete

I've noticed that the first warning indicates that no cloud has been selected but, as you identified in a previous issue from last year's Astricon repo, GKE generates k8s nodes with very long hostnames and includes the word "default" which is problematic for Kamailio. Netdiscover in best-effort mode is able to pick up the same public IP and private IP as if the environment variable was set to GCP. The only difference is in the hostname, when set-up to GCP it gets something such as gke--cluster-default-pool--..*****.internal and in best-effort ....bc.googleusercontent.com.

If I can provide any additional information to debug this issue, please let me know.

allow for unzipped config

I want to be able to construe /source directly from one or several configmaps / secrets without the need to package them as zip file previously. So that I can use kubernetes native (kustomize) gitops workflows (I could write my own kustomize zip generator, though) and factor configuration into layers through (kustomize) merge directives:

apiVersion: kustomize.config.k8s.io/v1beta1
kind: Kustomization
resources:
- ./00-rbac.yaml
- ./01-nats.yaml
- ./02-kamailio.yaml
- ./03-asterisk.yaml
secretGenerator:
- name: asterisk-config-base
  namespace: voip
  behavior: [create|replace|merge]  # this allows me to better factor configuration into pieces
  files:
  - modules.conf=asteriskconfig/modules.conf

- name: asterisk-config-ari.d
  namespace: voip
  behavior: [create|replace|merge]  # this allows me to better factor configuration into pieces
  files:
  - admin.conf=asteriskconfig/ari.d/admin.conf

- name: asterisk-config-pjsip.d
  namespace: voip
  behavior: [create|replace|merge]  # this allows me to better factor configuration into pieces
  files:
  - proxies.conf.tmpl=asteriskconfig/pjsip.d/proxies.conf.tmpl

- name: asterisk-config-extensions.d
  namespace: voip
  behavior: [create|replace|merge]  # this allows me to better factor configuration into pieces
  files:
  - inbound.conf.tmpl=asteriskconfig/extensions.d/inbound.conf.tmpl

    spec:
      volumes:
        - name: config
          emptyDir: {}
        - name: custom-base
          secret:
            secretName: asterisk-config-base
        - name: custom-ari-d
          secret:
            secretName: asterisk-config-ari.d
        - name: custom-pjsip-d
          secret:
            secretName: asterisk-config-pjsip.d
        - name: custom-extensions-d
          secret:
            secretName: asterisk-config-extensions.d
      containers:
        - name: config
          image: cycoresystems/asterisk-config
          volumeMounts:
            - name: config
              mountPath: /etc/asterisk
            - name: custom-base
              mountPath: /source
            - name: custom-arid
              mountPath: /source/ari.d
            - name: custom-pjsipd
              mountPath: /source/pjsip.d
            - name: custom-extensions-d
              mountPath: /source/extensions.d

Configuration users

hello, I would like to congratulate you on this project.
I have installed and configured but I have a problem with the test I would like to know how a user A can reach another user B with asterisk-config.

warning on startup for cli.conf

With the new CLI startup commands I now get a warning

[Mar 31 09:50:31] WARNING[1]: config.c:2020 process_text_line: No '=' (equal sign) in line 2 of /etc/asterisk/cli.conf

But according to the samples, your config is correct - https://github.com/asterisk/asterisk/blob/master/configs/samples/cli.conf.sample

Raising it just so its logged somewhere. Not much you can do about it

Asterisk Applications are not Loaded if Cloud is not Set

Hello,

I am currently trying to get this up and running on Amazon EKS and it is working for the most part. The only problem is if I do not specify a cloud, and run "core show applications" inside of the asterisk cli, these are the applications that I get:


              Answer: Answer a channel if ringing.
            BackGround: Play an audio file while waiting for digits of an extension to go to.
                Bridge: Bridge two channels.
            BridgeWait: Put a call into the holding bridge.
                  Busy: Indicate the Busy condition.
  CallCompletionCancel: Cancel call completion service
  CallCompletionRequest: Request call completion service for previous call
             ClearHash: Clear the keys from a specified hashname.
            ConfBridge: Conference bridge application.
            Congestion: Indicate the Congestion condition.
                  Dial: Attempt to connect to another device or endpoint and bridge the call.
             Directory: Provide directory of voicemail extensions.
            ExecIfTime: Conditional application execution based on the current time.
                 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.
             ImportVar: Import a variable from a channel into a new variable.
            Incomplete: Returns AST_PBX_INCOMPLETE value.
                   Log: Send arbitrary text to a selected log level.
         MailboxExists: Check to see if Voicemail mailbox exists.
           MessageSend: Send a text message.
                  MSet: Set channel variable(s) or function value(s).
           MusicOnHold: Play Music On Hold indefinitely.
                  NoOp: Do Nothing (No Operation).
              Playback: Play a file.
            Proceeding: Indicate proceeding.
              Progress: Indicate progress.
        RaiseException: Handle an exceptional condition.
             RetryDial: Place a call, retrying on failure allowing an optional exit extension.
                Return: Return from gosub routine.
               Ringing: Indicate ringing tone.
              SayAlpha: Say Alpha.
          SayAlphaCase: Say Alpha.
             SayDigits: Say Digits.
             SayNumber: Say Number.
           SayPhonetic: Say Phonetic.
                   Set: Set channel variable or function value.
           SetAMAFlags: Set the AMA Flags.
              StackPop: Remove one address from gosub stack.
      StartMusicOnHold: Play Music On Hold.
                Stasis: Invoke an external Stasis application.
       StopMusicOnHold: Stop playing Music On Hold.
               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.
      VoiceMailPlayMsg: Play a single voice mail msg from a mailbox by msg id.
                  Wait: Waits for some time.
             WaitDigit: Waits for a digit to be entered.
             WaitExten: Waits for an extension to be entered.
    -= 54 Applications Registered =-

asterisk-585c5c4ff4-6nqz5*CLI> core show applications
    -= Registered Asterisk Applications =-
              ADSIProg: Load Asterisk ADSI Scripts into phone
                AELSub: Launch subroutine built with AEL
                   AGI: Executes an AGI compliant application.
                Answer: Answer a channel if ringing.
           AudioSocket: Transmit and receive audio between channel and TCP socket
          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.
             BridgeAdd: Join a bridge that contains the specified channel.
            BridgeWait: Put a call into the holding bridge.
                  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.
             DAHDIScan: Scan DAHDI channels to monitor calls.
              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 media, 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.
               ForkCDR: Forks the current Call Data Record for this channel.
              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.
      HangupCauseClear: Clears hangup cause information from the channel that is available through HANGUPCAUSE.
                  ICES: Encode and stream using 'ices'.
             ImportVar: Import a variable from a channel into a new variable.
            Incomplete: Returns AST_PBX_INCOMPLETE value.
                   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.
           MessageSend: Send a text message.
             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.
                NBScat: Play an NBS local stream.
                 NoCDR: Tell Asterisk to not maintain a CDR for this channel.
                  NoOp: Do Nothing (No Operation).
             Originate: Originate a call.
                  Page: Page series of phones
          PauseMonitor: Pause monitoring of a channel.
                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.
        RaiseException: Handle an exceptional condition.
                  Read: Read a variable.
             ReadExten: Read an extension into a variable.
            ReceiveFAX: Receive a FAX and save as a TIFF/F file.
                Record: Record to a file.
              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.
          SayAlphaCase: 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: Sends a specified TIFF/F file as a FAX.
             SendImage: Sends an image file.
              SendText: Send a Text Message on a channel.
               SendURL: Send a URL.
                   Set: Set channel variable or function value.
           SetAMAFlags: Set the AMA Flags.
                   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.
                Stasis: Invoke an external Stasis application.
        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.
            StreamEcho: Echo media, up to 'N' streams of a type, and DTMF back to the calling party
                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.
             UserEvent: Send an arbitrary user-defined event to parties interested in a channel (AMI users and relevant res_stasis applications).
               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.
      VoiceMailPlayMsg: Play a single voice mail msg from a mailbox by msg id.
                  Wait: Waits for some time.
             WaitDigit: Waits for a digit to be entered.
             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.
             WaitUntil: Wait (sleep) until the current time is the given epoch.
                 While: Start a while loop.
            Zapateller: Block telemarketers with SIT.
    -= 152 Applications Registered =-

All of the Applications including the AudioSocket Application shows up. The problem is when the Audiosocket application is not registered when I do not specify the cloud, the call ends because Asterisk can not find the AudioSocket Application.

However, if I specify "aws" as the cloud, the asterisk-config container finds the wrong privateIPv4 address and puts it in the pjsip.d/k8s-asterisk-config.conf.tmpl file. So when the ACK comes from my provider, kamailio is unable to send the ACK back to the right nated ip address and the call hangs up after 32 seconds by asterisk because it never received the ACK.

I have solved this by 1) not providing it a cloud parameter. and 2) ssh into the asterisk container, running netdiscover or "hostname -I" to get the containers privateIPv4 and setting it in the pjsip.d/k8s-asterisk-config.conf.tmpl.

I would like to know, what is the best solution to get both the right ip address and to get the asterisk Audiosocket application to register?

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